Re: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller
Doug: for what it's worth I am having the exact same nightmare. Not sure exactly when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers are broken. All legs of the call are dropped when the xfer is executed. A calls B, B xfer to C and (C) blips for a split second like its ringing but then all calls go dead. I tried to debug myself using some sip tracing but I didn't get very far. I even tried mucking around with a few settings in my Polycom provisioning I thought might be related e.g. voIpProt.SIP.allowTransferOnProceeding voIpProt.SIP.connectionReuse.useAlias voIpProt.SIP.useContactInReferTo voIpProt.SIP.conference.parallelRefer voIpProt.SIP.strictLineSeize voIpProt.SIP.strictUserValidation voIpProt.SIP.strictReplacesHeader voIpProt.SIP.useContactInReferTo and also upgraded to the new 3.3.4 firmware which is out yesterday, didn't change a thing. stuck here for now, Attended xfers seem to work.I am not sure this is a Polycom-specific issue because I was seeing this bad behavior even using some Softphones I set up for testing. my next recourse is to try rolling back to 1.8.8.0 or earlier and if that fixes it then I will open a JIRA ticket with more details. Luke -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Thursday, January 05, 2012 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in we answer transfer, everything works fine. But if we call out to a customer then transfer to another internal extension, that extension quickly rings then the call is immediately gone hung up. We are using Polycom firmware 3.3.3. In troubleshooting this analyzing the asterisk logs ( asterisk SIP debug), I am seeing a few interesting items. Any help would be appreciated. [...] Thanks, - Doug Mortensen -- _ -- 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] Speech recognition in asterisk using google voice API
On 01/07/2012 09:34 AM, Bruce B wrote: Added two new features to the script: Timeout value and speechdata type. *exten = s,n,agi(speech-recog.agi,en-US,3000,phoneNumb)* - Will listen for 3 seconds and sanitize return as a single number without any spaces in between. This helps when one reads phone number in format 415-554-2323 and google returns, 415 554 2323 as result which is not very usable. *exten = s,n,agi(speech-recog.agi,en-US,2,string)* - Will listen for 20 second and return result as provided by Google untouched. It would be great to see them in future versions as I seem to need them dearly in a real life scenario. Updated script attached. -Bruce Thank you Bruce for the testing and the suggestions. Both features added in the script. Timeout can now be set by the user, also -1 means no timeout and the recording keeps going till # is pressed. Space gets stripped between digits, this is now the default behavior and there's no need to determine the 'speechdata' type. The updated code can be found here: https://github.com/zaf/asterisk-speech-recog/tarball/master Next on my TODO list is to make use of the asterisk speech recognition API (https://wiki.asterisk.org/wiki/display/AST/Speech+Recognition+API) This will make the application actually usable for real case scenarios and not a proof of concept as it is now. 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] Couple of questions: SIP ALG, allowguest=no
Hello I just read this article about an Asterisk server that got hacked to make free international calls through an ITSP: www.rowetel.com/blog/?p=2210 I have a couple of questions: 1. Am I correct in understanding that SIP ALG on a router makes it easier to host an Asterisk server on a private LAN behind a NAT router (no need to map ports for RTP + outgoing packets can be sent directly to the remote SIP client instead of going through the Asterisk server to rewrite the RTP port numbers)? www.voip-info.org/wiki/view/Routers+SIP+ALG 2. If allowguest=no is commented out, it means that any SIP client on the Net can connect to the Asterisk server and make outgoing calls like legitimate SIP clients? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] public ip issue with asterisk cluster
Hi, I have an Opensips server dispatching to 3 Asterisk servers. I would like to assign public IPs to all of these servers and avoid NAT altogether - phones will also have public IPs. The way I set this in the lab, all the SIP traffic goes thru the SIP proxy (Opensips) and RTP goes directly between the Asterisk servers and the UAs. The issue is that our provider (they will be both sip trunk and internet access provider for us) wants to assign us only 1 public IP on their voice network - they are saying that the above design is unusual. I'm new to this, is it? If we end up getting only 1 public IP, I assume putting all behind NAT (or assigning the public IP to opensips and putting the asterisk servers behind NAT) will do it. rtpproxy is also setup on the Opensips server just in case - I can use it to force the RTP traffic thru the sip proxy. Any other way? All I want to do is load balance the RTP traffic, avoid any unnecessay processing and bottlenecks (rtpproxy, etc.). Any thoughts? Thanks, Matt -- _ -- 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] best softphone for 2012?
On 01/06/2012 05:00 PM, Tom Poe wrote: Just installed asterisknow 1.6. I can access freepbx. I need to test system on my LAN. Which softphone is best to use? I'm running ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for incoming/outgoing calls. No video. Tom We came down to linphone and qutecom. linphone echo canceller was a modest plus. But what really made us choose linphone was you use it on android/iphone. That has been a huge plus. As a bonus, you can use any degegistered smartphone - that is, one not hooked up to the cellular network,only wireless - as a softphone. sean -- _ -- 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] best softphone for 2012?
On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com wrote: But what really made us choose linphone was you use it on android/iphone. That has been a huge plus. As a bonus, you can use any degegistered smartphone - that is, one not hooked up to the cellular network,only wireless - as a softphone. I guess you meant de-registered smartphone : what does it mean? -- _ -- 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 10.0 1.4 - iax codec are not compatible
- Original Message - I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have 11, digest has pstn-1270 NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device KMIEC Z sip:7804715665@10.0.0.110;tag=1c1222950155 Anybody know what is the magic solution to get a CallerID to work? In addition iax -codec are not compatible with earlier asterisk (1.4); I have selected ulaw / alaw but Asterisk 1.4 wants GSM: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. IAX is a signalling protocol, not a codec. And, interop between the various branches of Asterisk is not a problem. Looking at your logs, the problem appears to be you have a codec mismatch between peers, or your authentication details are wrong. Please also check that you have calltokens set the same on both sides (enabled on both or disabled on both). --Tim -- _ -- 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] Blind transfers being cancelled by asterisk hanging up on remote caller
On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg l...@solvent-llc.com wrote: Doug: for what it's worth I am having the exact same nightmare. Not sure exactly when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers are broken. All legs of the call are dropped when the xfer is executed. A calls B, B xfer to C and (C) blips for a split second like its ringing but then all calls go dead. I tried to debug myself using some sip tracing but I didn't get very far. I even tried mucking around with a few settings in my Polycom provisioning I thought might be related e.g. voIpProt.SIP.allowTransferOnProceeding voIpProt.SIP.connectionReuse.useAlias voIpProt.SIP.useContactInReferTo voIpProt.SIP.conference.parallelRefer voIpProt.SIP.strictLineSeize voIpProt.SIP.strictUserValidation voIpProt.SIP.strictReplacesHeader voIpProt.SIP.useContactInReferTo and also upgraded to the new 3.3.4 firmware which is out yesterday, didn't change a thing. stuck here for now, Attended xfers seem to work.I am not sure this is a Polycom-specific issue because I was seeing this bad behavior even using some Softphones I set up for testing. my next recourse is to try rolling back to 1.8.8.0 or earlier and if that fixes it then I will open a JIRA ticket with more details. Luke -- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Thursday, January 05, 2012 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in we answer transfer, everything works fine. But if we call out to a customer then transfer to another internal extension, that extension quickly rings then the call is immediately gone hung up. We are using Polycom firmware 3.3.3. In troubleshooting this analyzing the asterisk logs ( asterisk SIP debug), I am seeing a few interesting items. Any help would be appreciated. [...] Thanks, - Doug Mortensen I can't reproduce this on a test system with Asterisk 1.8.8.1 using a Polycom 335 and 550 running firmware 3.2.6. I called an external number using Vitelity then blind transferred to the other phone. I am interested as I have a production system with Polycom 335 phones running 1.8.7.0 that works. Ryan -- _ -- 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] best softphone for 2012?
On Fri, 2012-01-06 at 16:00 -0600, Tom Poe wrote: Just installed asterisknow 1.6. I can access freepbx. I need to test system on my LAN. Which softphone is best to use? I'm running ubuntu on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX for incoming/outgoing calls. No video. Tom Hi, Our requirements were different, so we came to three candidates; linphone, ekiga and jitsi Linhphone is easy to pre-configure from a script and the buttons are easier to use, but lacks the possility for an ldap-adres-book. With ekiga you have the adresbook, but you have to use the mouse everywhere (the return-button gives unexpected results) And with jitsi (java-based) you are independant of Qt/GTK. hw -- _ -- 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 10.0 1.4 - iax codec are not compatible
On 01/07/12 08:50, Tim Nelson wrote: - Original Message - I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have 11, digest has pstn-1270 NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device KMIEC Z sip:7804715665@10.0.0.110;tag=1c1222950155 Anybody know what is the magic solution to get a CallerID to work? In addition iax -codec are not compatible with earlier asterisk (1.4); I have selected ulaw / alaw but Asterisk 1.4 wants GSM: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. IAX is a signalling protocol, not a codec. And, interop between the various branches of Asterisk is not a problem. Looking at your logs, the problem appears to be you have a codec mismatch between peers, or your authentication details are wrong. Please also check that you have calltokens set the same on both sides (enabled on both or disabled on both). --Tim I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. How about incoming CallerID from PSTN, why isn't asterik 1.8 nor 10.0 working correctly? I'm not getting any caller ID when I tried both versions 1.8 nor 10.0 All I'm getting is a WARNING: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have 11, digest has pstn-1270 NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device KMIEC Z sip:7804715665@10.0.0.110;tag=1c1222950155 Incoming CallerID from PSTN in Asterisk 1.4.39 is working perfectly. -- Joseph -- _ -- 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 10.0 1.4 - iax codec are not compatible
In article 20120107163819.gc3...@syscon7.inet, Joseph syscon...@gmail.com wrote: I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. Check also the allow= and disallow= directives in the [general] section of your iax.conf. It may be that the call is not matching the friend section you think it is. Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] Asterisk 10.0 1.4 - iax codec are not compatible
Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. According to this log, server 192.168.141.8 has codecs defined as 0xc (ulaw and alaw), which matches your config, but the other end has codecs 0x703 (g723, gsm, g729, speex, ilbc) which does not match your config. You should debug and make sure the call setup is choosing the peers you are expecting in your config. -- Technical Support http://www.telesip.net -- _ -- 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] best softphone for 2012?
On Sat, Jan 7, 2012 at 9:34 AM, Gilles codecompl...@free.fr wrote: On Sat, 07 Jan 2012 09:27:29 -0500, sean darcy seandar...@gmail.com wrote: But what really made us choose linphone was you use it on android/iphone. That has been a huge plus. As a bonus, you can use any degegistered smartphone - that is, one not hooked up to the cellular network,only wireless - as a softphone. I guess you meant de-registered smartphone : what does it mean? Yes, I did mean de-registered. I meant a phone that no longer has the ability to use the cellular network - only wifi. For instance, we have a couple of Droids that used to be on Verizon. They work just fine as sip-phones over wifi. sean -- _ -- 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 - AGI (perl) - sqlplus(oracle)
On Fri, 6 Jan 2012, Dale Noll wrote:
I found the following lines to be helpful.
$ENV{TNS_ADMIN}=/usr/lib/oracle/11.2/client/;
$ENV{ORACLE_HOME}=/usr/lib/oracle/11.2/client/;
$ENV{LD_LIBRARY_PATH}=/usr/lib/oracle/11.2/client/lib/;
I think a 'better practice' would be to put the 'stuff likely to change'
into the environment variables of the Asterisk process so they will
'trickle down' to sub-processes like AGIs.
This way, when you upgrade Oracle, you don't have to track down and change
all affected AGIs.
Something like this snippet from my Asterisk start up script:
nice --adjustment=-20\
env --ignore-environment\
HOSTNAME=${HOSTNAME}\
LD_LIBRARY_PATH='/usr/lib/oracle/11.2/client/lib/'\
ORACLE_HOME='/usr/lib/oracle/11.2/client/'\
PATH=${PATH}\
TNS_ADMIN='/usr/lib/oracle/11.2/client/'\
$ASTERISK $START_OPTIONS
I like to 'ignore' the environment of the process executing the script
that starts Asterisk and add in only what is needed -- I'm a 'parts left
out don't get broken' kind of guy :)
Can you give this a try and report back?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] best softphone for 2012?
On Sat, 7 Jan 2012 12:34:44 -0500, Sean Darcy seandar...@gmail.com wrote: Yes, I did mean de-registered. I meant a phone that no longer has the ability to use the cellular network - only wifi. For instance, we have a couple of Droids that used to be on Verizon. They work just fine as sip-phones over wifi. Thanks. -- _ -- 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 10.0 1.4 - iax codec are not compatible
Chances are the incoming call is not matching anything in iax.conf. turn on iax debug, try a call, post the results. Maybe someone familiar with IAX can help you. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres Sent: Saturday, January 07, 2012 12:20 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 10.0 1.4 - iax codec are not compatible Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. According to this log, server 192.168.141.8 has codecs defined as 0xc (ulaw and alaw), which matches your config, but the other end has codecs 0x703 (g723, gsm, g729, speex, ilbc) which does not match your config. You should debug and make sure the call setup is choosing the peers you are expecting in your config. -- Technical Support http://www.telesip.net -- _ -- 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] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible
On 01/07/12 17:13, Tony Mountifield wrote: In article 20120107163819.gc3...@syscon7.inet, Joseph syscon...@gmail.com wrote: I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. Check also the allow= and disallow= directives in the [general] section of your iax.conf. It may be that the call is not matching the friend section you think it is. Tony Thanks Tony, that was it and it solved BOTH problem codec and callerID Changing in Asterisk 1.8.7 [general] section allow=all Thank again Tony! My [general] section I had by default: ;allow=all ; same as bandwidth=high ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ;allow=gsm ; Always allow GSM, it's cool :) -- Joseph -- _ -- 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] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible
This means you are allowing guest calls. A VERY bad thing. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Saturday, January 07, 2012 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible On 01/07/12 17:13, Tony Mountifield wrote: In article 20120107163819.gc3...@syscon7.inet, Joseph syscon...@gmail.com wrote: I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. Check also the allow= and disallow= directives in the [general] section of your iax.conf. It may be that the call is not matching the friend section you think it is. Tony Thanks Tony, that was it and it solved BOTH problem codec and callerID Changing in Asterisk 1.8.7 [general] section allow=all Thank again Tony! My [general] section I had by default: ;allow=all ; same as bandwidth=high ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ;allow=gsm ; Always allow GSM, it's cool :) -- Joseph -- _ -- 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] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible
On 01/07/12 13:27, Eric Wieling wrote: This means you are allowing guest calls. A VERY bad thing. Doesn't it pertain to codes only? in my [guest] section I have: ;[guest] ;type=user ;context=default ;callerid=Guest IAX User so it is disabled, isn't it? -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Saturday, January 07, 2012 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible On 01/07/12 17:13, Tony Mountifield wrote: In article 20120107163819.gc3...@syscon7.inet, Joseph syscon...@gmail.com wrote: I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. Check also the allow= and disallow= directives in the [general] section of your iax.conf. It may be that the call is not matching the friend section you think it is. Tony Thanks Tony, that was it and it solved BOTH problem codec and callerID Changing in Asterisk 1.8.7 [general] section allow=all Thank again Tony! My [general] section I had by default: ;allow=all ; same as bandwidth=high ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ;allow=gsm ; Always allow GSM, it's cool :) -- Joseph -- _ -- 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] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible
The codecs and contexts defined in [general] apply to unauthenticated calls. If the incoming call matched the entry in sip.conf or iax.conf then the codecs in that entry would be used. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Saturday, January 07, 2012 1:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible On 01/07/12 13:27, Eric Wieling wrote: This means you are allowing guest calls. A VERY bad thing. Doesn't it pertain to codes only? in my [guest] section I have: ;[guest] ;type=user ;context=default ;callerid=Guest IAX User so it is disabled, isn't it? -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Saturday, January 07, 2012 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible On 01/07/12 17:13, Tony Mountifield wrote: In article 20120107163819.gc3...@syscon7.inet, Joseph syscon...@gmail.com wrote: I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. Check also the allow= and disallow= directives in the [general] section of your iax.conf. It may be that the call is not matching the friend section you think it is. Tony Thanks Tony, that was it and it solved BOTH problem codec and callerID Changing in Asterisk 1.8.7 [general] section allow=all Thank again Tony! My [general] section I had by default: ;allow=all ; same as bandwidth=high ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ;allow=gsm ; Always allow GSM, it's cool :) -- Joseph -- _ -- 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] Blind transfers being cancelled by asterisk hanging up on remote caller
Could be the phone firmware. I'm not sure. I'll probably get it resolved next week post back how it goes. - Doug Mortensen Sent via DroidX2 on Verizon Wireless™ -Original message- From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sat, Jan 7, 2012 15:59:36 GMT+00:00 Subject: Re: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg l...@solvent-llc.commailto:l...@solvent-llc.com wrote: Doug: for what it's worth I am having the exact same nightmare. Not sure exactly when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers are broken. All legs of the call are dropped when the xfer is executed. A calls B, B xfer to C and (C) blips for a split second like its ringing but then all calls go dead. I tried to debug myself using some sip tracing but I didn't get very far. I even tried mucking around with a few settings in my Polycom provisioning I thought might be related e.g. voIpProt.SIP.allowTransferOnProceeding voIpProt.SIP.connectionReuse.useAlias voIpProt.SIP.useContactInReferTo voIpProt.SIP.conference.parallelRefer voIpProt.SIP.strictLineSeize voIpProt.SIP.strictUserValidation voIpProt.SIP.strictReplacesHeader voIpProt.SIP.useContactInReferTo and also upgraded to the new 3.3.4 firmware which is out yesterday, didn't change a thing. stuck here for now, Attended xfers seem to work.I am not sure this is a Polycom-specific issue because I was seeing this bad behavior even using some Softphones I set up for testing. my next recourse is to try rolling back to 1.8.8.0 or earlier and if that fixes it then I will open a JIRA ticket with more details. Luke -- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Thursday, January 05, 2012 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in we answer transfer, everything works fine. But if we call out to a customer then transfer to another internal extension, that extension quickly rings then the call is immediately gone hung up. We are using Polycom firmware 3.3.3. In troubleshooting this analyzing the asterisk logs ( asterisk SIP debug), I am seeing a few interesting items. Any help would be appreciated. [...] Thanks, - Doug Mortensen I can't reproduce this on a test system with Asterisk 1.8.8.1 using a Polycom 335 and 550 running firmware 3.2.6. I called an external number using Vitelity then blind transferred to the other phone. I am interested as I have a production system with Polycom 335 phones running 1.8.7.0 that works. Ryan -- _ -- 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] Blind transfers being cancelled by asterisk hanging up on remote caller
Oh crap. I just reread the previous post realized I'm not alone. Hallelujah! I'll post back more info soon. - Doug Mortensen Sent via DroidX2 on Verizon Wireless™ -Original message- From: Ryan Wagoner rswago...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sat, Jan 7, 2012 15:59:36 GMT+00:00 Subject: Re: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller On Sat, Jan 7, 2012 at 5:19 AM, Luke Hamburg l...@solvent-llc.commailto:l...@solvent-llc.com wrote: Doug: for what it's worth I am having the exact same nightmare. Not sure exactly when it started but I believe it was a change in 1.8.8.1 / 1.8.9.0-rc1 (I am running 1.8.9rc1). I also have Polycom (335, 550, 650) and blind transfers are broken. All legs of the call are dropped when the xfer is executed. A calls B, B xfer to C and (C) blips for a split second like its ringing but then all calls go dead. I tried to debug myself using some sip tracing but I didn't get very far. I even tried mucking around with a few settings in my Polycom provisioning I thought might be related e.g. voIpProt.SIP.allowTransferOnProceeding voIpProt.SIP.connectionReuse.useAlias voIpProt.SIP.useContactInReferTo voIpProt.SIP.conference.parallelRefer voIpProt.SIP.strictLineSeize voIpProt.SIP.strictUserValidation voIpProt.SIP.strictReplacesHeader voIpProt.SIP.useContactInReferTo and also upgraded to the new 3.3.4 firmware which is out yesterday, didn't change a thing. stuck here for now, Attended xfers seem to work.I am not sure this is a Polycom-specific issue because I was seeing this bad behavior even using some Softphones I set up for testing. my next recourse is to try rolling back to 1.8.8.0 or earlier and if that fixes it then I will open a JIRA ticket with more details. Luke -- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Thursday, January 05, 2012 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Blind transfers being cancelled by asterisk hanging up on remote caller Hello all, I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that blindpreferred=1 (all transfers default as blind transfers). If a customer calls in we answer transfer, everything works fine. But if we call out to a customer then transfer to another internal extension, that extension quickly rings then the call is immediately gone hung up. We are using Polycom firmware 3.3.3. In troubleshooting this analyzing the asterisk logs ( asterisk SIP debug), I am seeing a few interesting items. Any help would be appreciated. [...] Thanks, - Doug Mortensen I can't reproduce this on a test system with Asterisk 1.8.8.1 using a Polycom 335 and 550 running firmware 3.2.6. I called an external number using Vitelity then blind transferred to the other phone. I am interested as I have a production system with Polycom 335 phones running 1.8.7.0 that works. Ryan -- _ -- 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] [SOLVED - need explanation] Asterisk 10.0 1.4 - iax codec are not compatible
On 01/07/12 13:42, Eric Wieling wrote: The codecs and contexts defined in [general] apply to unauthenticated calls. If the incoming call matched the entry in sip.conf or iax.conf then the codecs in that entry would be used. I just change in iax.conf in [general] section: from: allow=all to: allow=ulaw allow=alaw Codec and with Asterisk 1.4.39 is working but CallerID is working as well. Can someone explain to me how changing in allow=ulaw/alaw in iax.conf effect the display of incoming CallerID from PSTN line. My AudioCode gateway communicate with asterisk server using SIP, so why changed to iax.conf affect sip communication? So how did it happen? It is good, it is working but can someone with more knowledge explain us WHY? As I'm sure I'll not be the only one with this problem of CallerID. -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Saturday, January 07, 2012 1:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible On 01/07/12 13:27, Eric Wieling wrote: This means you are allowing guest calls. A VERY bad thing. Doesn't it pertain to codes only? in my [guest] section I have: ;[guest] ;type=user ;context=default ;callerid=Guest IAX User so it is disabled, isn't it? -- Joseph -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Saturday, January 07, 2012 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [SOLVED] Asterisk 10.0 1.4 - iax codec are not compatible On 01/07/12 17:13, Tony Mountifield wrote: In article 20120107163819.gc3...@syscon7.inet, Joseph syscon...@gmail.com wrote: I'm not sure this is the case: Asterisk-1.4.39 [home_server] type=friend host=dynamic secret=123456 context=extensions disallow=all allow=ulaw allow=alaw requirecalltoken=no Asterisk-1.8.7 [clinic_server] type=friend host=dynamic context=internal disallow=all allow=ulaw allow=alaw requirecalltoken=no Error message on asterisk-1.8.7 chan_iax2.c:10666 socket_process: Call rejected by 192.168.141.1: Unable to negotiate codec Error message on asterisk-1.4.39 chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. Check also the allow= and disallow= directives in the [general] section of your iax.conf. It may be that the call is not matching the friend section you think it is. Tony Thanks Tony, that was it and it solved BOTH problem codec and callerID Changing in Asterisk 1.8.7 [general] section allow=all Thank again Tony! My [general] section I had by default: ;allow=all ; same as bandwidth=high ;disallow=g723.1; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. ;allow=gsm ; Always allow GSM, it's cool :) -- Joseph -- _ -- 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
