Re: [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe
On Tue, 22 Feb 2011 16:33:05 -0800 (PST), Steve Edwards asterisk@sedwards.com wrote: While the documentation on the protocol is clear, nobody gets it right the first time -- which is why I always suggest using an established library for the language of your choice. Indeed, neither the 2nd nor the 3rd edition of the Asterisk book make it clear that an AGI script _must_ read all data from stdin before going ahead. Since it's a pretty simple protocol, it doesn't look like there's a Lua library to handle AGI scripts. 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
Re: [asterisk-users] calls between iax and sip
Thanks steve for your response the details is below When i call from iax extension (1018) to sip extension there is no issue == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629 currently running on srvradio (pid = 24818) Verbosity is at least 3 -- Accepting UNAUTHENTICATED call from 192.168.5.131: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (alaw|ulaw), priority = mine -- Executing [MCALL106^1298455141.287500@agents:1] Set(IAX2/1018-6, AH_TEMP=106^1298455141.287500) in new stack -- Executing [MCALL106^1298455141.287500@agents:2] NoOp(IAX2/1018-6, [106^1298455141.287500]) in new stack -- Executing [MCALL106^1298455141.287500@agents:3] Set(IAX2/1018-6, AH_EXTEN=106) in new stack -- Executing [MCALL106^1298455141.287500@agents:4] Set(IAX2/1018-6, AHEEVA_TRACKNUM=1298455141.287500) in new stack -- Executing [MCALL106^1298455141.287500@agents:5] Goto(IAX2/1018-6, agents|106|1) in new stack -- Goto (agents,106,1) -- Executing [106@agents:1] Dial(IAX2/1018-6, SIP/106) in new stack -- Called 106 -- SIP/106-095133e8 is ringing -- SIP/106-095133e8 answered IAX2/1018-6 == Agent '1018' logged out == Spawn extension (agents, AH1018, 1) exited non-zero on 'IAX2/1018-4' == Spawn extension (agents, 106, 1) exited non-zero on 'IAX2/1018-6' -- Executing [h@agents:1] GotoIf(IAX2/1018-4, 0?3:2) in new stack -- Executing [h@agents:1] GotoIf(IAX2/1018-6, 1?3:2) in new stack -- Goto (agents,h,2) -- Executing [h@agents:2] AHEventsProxy(IAX2/1018-4, MSG_TYPE_TERMINATE_CALL1298455155) in new stack AHEventsProxy: Channel [IAX2/1018-4]. Data [MSG_TYPE_TERMINATE_CALL1298455155] -- chan is IAX2/1018-4 AHEventsProxy: Send To CtiServer: socket:[67]. message:[41,1298455155Ipbx01^~] -- Executing [h@agents:3] Hangup(IAX2/1018-4, ) in new stack == Spawn extension (agents, h, 3) exited non-zero on 'IAX2/1018-4' -- Hungup 'IAX2/1018-4' -- Goto (agents,h,3) -- Executing [h@agents:3] Hangup(IAX2/1018-6, ) in new stack == Spawn extension (agents, h, 3) exited non-zero on 'IAX2/1018-6' -- Hungup 'IAX2/1018-6' -- Accepting UNAUTHENTICATED call from 192.168.5.131: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (alaw|ulaw), priority = mine -- Executing [AH1018@agents:1] AgentLogin(IAX2/1018-9, 1018|s) in new stack -- Started music on hold, class 'none', on channel 'IAX2/1018-9' == Agent '1018' logged in (format ulaw/slin) -- Stopped music on hold on IAX2/1018-9 [Feb 23 09:59:22] NOTICE[25420]: chan_sip.c:15012 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 106 srvradio*CLI but when i call from sip extension 106 to iax extension (1018) i got the message below == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629 currently running on srvradio (pid = 24818) Verbosity is at least 3 [Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite: Call from '106' to extension '1018' rejected because extension not found. srvradio*CLI thank you for your help 2011/2/22 Danny Nicholas da...@debsinc.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, February 22, 2011 12:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] calls between iax and sip On Tue, 22 Feb 2011, salaheddine elharit wrote: i have asterisk installed and i have configured a client iax and sip without any issue, when i call a internal extension sip from iax there is no problem but when i call a iax extension from sip extension the result is KO(wrong number) any help please No details, no help. Crank up verbosity on the CLI and see if the messages yield a clue. If not, please post the console messages. Isn't Dionne Warrick a poster on this list? :) -- _ -- 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:
[asterisk-users] extend the timout on ringing for pri or sip
Hi Does anyone know how i could extend the timer for the ringing time on a pri or sip trunk ? Today the call gets a cancel request after a minute if not answerd yet is it on asterisk or is a provider side setting? -- _ -- 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] AMI FullyBooted issue
Hi We're still testing out asterisk 1.8 (using 1.8.2.2 from rpm package) before putting it into production and I'm observing an odd issue when using the AMI Every request I send to the AMI just results in a FullyBooted response rather than the expected response. Here are some examples from my logs -- Call started: 22/02/2011 11:34:03 -- action: command command: core show channels Event: FullyBooted Privilege: system,all SequenceNumber: 1706 File: manager.c Line: 2937 Func: action_login Status: Fully Booted -- Call started: 22/02/2011 10:28:15 -- action: command command: sip show peers Event: FullyBooted Privilege: system,all SequenceNumber: 1610 File: manager.c Line: 2937 Func: action_login Status: Fully Booted Has anyone else experienced anything like this? -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Adhearsion 1.0.1 Released
The Adhearsion team announces the release of Adhearsion version 1.0.1. Adhearsion is an open source Ruby-language framework for creating telephony applications. This update primarily addresses compatibility with newer versions of other software but also adds native support for Bundler to newly created Adhearsion applications. Here are some highlights from the changelog: Handling of new Asterisk 1.6/1.8 events Improved control of Asterisk Queues Two new dialplan methods have been added: say_chars and say_phonetic Ruby 1.9 is now an officially supported platform Fix compatibility with Rails 3 Bundler now included by default for new Adhearsion applications Not bad for a dot release! You can read the full CHANGELOG here. As always I'd like to thank the Adhearsion community for their contributions to this release. Special thanks to contributors Ben Langfeld, Robert Jackson and Matthew Clark. To install Adhearsion just type gem install adhearsion at your nearest command prompt. For help getting started, checkout our Wiki and Getting Started pages. As always, you can find us on irc.freenode.net #adhearsion or our Google Groups mailing list. Contributors welcome! Check out the sources on Adhearsion's Github. /BAK/ -- Ben Klang bkl...@mojolingo.com 404.475.4841 Mojo Lingo -- Voice applications that work like magic http://mojolingo.com Twitter: @mojolingo-- _ -- 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] AMI FullyBooted issue
On 11-02-23 05:39 AM, Ishfaq Malik wrote: Has anyone else experienced anything like this? There is a patch on the issue tracker[1], please test it out and report your feedback. [1] https://issues.asterisk.org/view.php?id=18168 -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card
This is the closest thing I was able to find in my wctdm.c file:
if ((blah 0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors when compling.
On Tue, Feb 22, 2011 at 11:43 PM, Shaun Ruffell sruff...@digium.com wrote:
On Tue, Feb 22, 2011 at 11:00:22PM -0500, C F wrote:
On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell sruff...@digium.com wrote:
On 2/21/11 4:46 PM, C F wrote:
I just installed an FXS module onto a 4 channel tdm thats about 5
years old and it wont work. Running dmesg I can see the following
error:
[snip]
! Init Indirect Registers UNSUCCESSFULLY.
Indirect Registers failed verification.
Module 0: FAILED FXS (FCC)
Module 1: Installed -- AUTO FXO (FCC mode)
Module 2: Installed -- AUTO FXO (FCC mode)
Module 3: Installed -- AUTO FXO (FCC mode)
Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)
Does this have to do with the fact that the module is way newer than the
card?
Not having much direct experience with the wctdm.c driver, that would be my
guess. You might be able to edit the wctdm_proslic_insane() function to
force the FLAG_3215 on for the card and see if that gives you a different
result.
How/Where would I do that?
Around line 1297 of drivers/dahdi/wctdm.c you could change:
if (wctdm_getreg(wc, card, 1) 0x80)
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
to
wc-flags[card] |= FLAG_3215;
and just skip the read of register 1. I don't know if this will work in your
case, but it's something to try.
--
Shaun Ruffell
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] AMI FullyBooted issue
On Wed, 2011-02-23 at 09:37 -0500, Paul Belanger wrote: On 11-02-23 05:39 AM, Ishfaq Malik wrote: Has anyone else experienced anything like this? There is a patch on the issue tracker[1], please test it out and report your feedback. [1] https://issues.asterisk.org/view.php?id=18168 Hi Paul Thanks for that. Unfortunately I'm on strict instructions to use rpm packages only so can't test your patch. However, the eventfilter work around solves my problem so thanks for bringing it to my attention. Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS:CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf - start ... register = john:j0nhp...@66.128.xx.xxx ... [john-peer] type=peer defaultuser=john remotesecret=j0nhp4ss ;secret=j0nhp4ss host=66.128.XX.XXX directmedia=no dtmfmode=rfc2833 context=jonh-context - end When I send a call to that block, I receive the following response unless (I explicitly indicate a 'secret' parameter, no matter if 'remotesecret' parameter was indicated): Forbidden from 'Test Account sip:9_...@66.128.xx.xxx;tag=as749a7ced' If I set the 'secret' parameter, everything goes smoothly as expected. Maybe I'm obviating something 'basic', but the CHANGES file says: - Added a new configuration option remotesecret for authentication to remote services. For backwards compatibility, secret still has the same function as before, but now you can configure both a remote secret and a local secret for mutual authentication. - and on sip.conf.sample - ;remotesecret=guessit ; Our password to their service - I thought that 'remotesecret' is used to authenticate myself when placing a call to the remote network, as I used to do with 'secret' parameter. Doing a: grep -ir 'remotesecret' . (inside the Asterisk source directory) indicates that only this files mention that parameter: ./ChangeLog ./channels/chan_sip.c ./CHANGES: ./configs/sip.conf.sample Could someone please point me to documentation regarding this two parameters? Thanks in advice. -- Jose P. Espinal http://www.eslackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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] REFER and dialplan broken (as documented in chan_sip.c on line 11951)
There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I’m quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren’t satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Mea Culpa - I see this in my 1.4.37 source as well (line 8401 in this release chan_sip.c). Hopefully someone like Tilghman will address this; a simple hack would be to create a C daemon that did a core show channels and transmit to appropriate results back for referral. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I’m quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren’t satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I use Polycom 501's and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] calls between iax and sip
On Wed, 23 Feb 2011, salaheddine elharit wrote: == Agent '1018' logged in (format ulaw/slin) An agent is not the same as an extension. but when i call from sip extension 106 to iax extension (1018) i got the message below [Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite: Call from '106' to extension '1018' rejected because extension not found. This NOTICE is from the SIP channel driver, not the IAX channel driver. What does the dial statement that generates the above NOTICE look like? What does 'iax2 show peer 1018' display? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501’s and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I’m quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren’t satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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 -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Do you use the Queue command natively or from the AGI? In the example you gave, if you did a core show channels, I assume that Agent007 would be idle, but ineligible for Queue activity. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501's and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote: Do you use the Queue command “natively” or from the AGI? In the example you gave, if you did a “core show channels”, I assume that Agent007 would be idle, but ineligible for Queue activity. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501’s and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I’m quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren’t satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
My bad - natively means using the Queue command from the dialplan. Since the powers that be are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote: Do you use the Queue command natively or from the AGI? In the example you gave, if you did a core show channels, I assume that Agent007 would be idle, but ineligible for Queue activity. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501's and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I'm quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren't satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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 --
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.com wrote: My bad – “natively” means using the Queue command from the dialplan. Since the “powers that be” are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:57 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote: Do you use the Queue command “natively” or from the AGI? In the example you gave, if you did a “core show channels”, I assume that Agent007 would be idle, but ineligible for Queue activity. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501’s and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I’m quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren’t satisfied with the progress being made, you should research your own solution and/or offer a bounty. --
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Un-top-posting... On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.com wrote: My bad – “natively” means using the Queue command from the dialplan. Since the “powers that be” are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. On Wed, 23 Feb 2011, vip killa wrote: Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit And what are you 'in it' for? The developer community is populated by many kinds of people. Some do it because it's their job, some do it for the challenge, some do it 'for the greater good' and some do it as 'a gun for hire.' Whatever their motivation, are you receiving more than you give? My guess is 'yes' which makes it 'fortunate' for you and me. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Implying that the Asterisk developers (which is itself a fairly nebulous statement since those who contribute to Asterisk are many and come from different companies/countries/etc.) are not in it to make a good product but to make a profit is not only highly insulting but a complete mischaracterization of what you were told on IRC. What you were told was that there are essentially three choices (and this goes for pretty much any open source software, as already stated). You may either fix it yourself (if you have the skills), pay someone to fix it for you (if you can or must trade money for expediency), or wait for someone else with the skills and/or money necessary to fix it. Regards, - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: My bad - natively means using the Queue command from the dialplan. Since the powers that be are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. 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 vip killa Sent: Wednesday, February 23, 2011 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: Do you use the Queue command natively or from the AGI? In the example you gave, if you did a core show channels, I assume that Agent007 would be idle, but ineligible for Queue activity. 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 vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: I use Polycom 501's and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? 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 vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 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 vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
It's simple, if a product is broken shouldn't it be fixed? In this case the answer is for a price which is absurd because it is an open source product. If there was a decent community of developers surrounding this open source project, it would be fixed simply because it's broken, no questions asked. On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley bradley.watk...@compuware.com wrote: Implying that the Asterisk developers (which is itself a fairly nebulous statement since those who contribute to Asterisk are many and come from different companies/countries/etc.) are “not in it to make a good product” but to make a “profit” is not only highly insulting but a complete mischaracterization of what you were told on IRC. What you were told was that there are essentially three choices (and this goes for pretty much any open source software, as already stated). You may either fix it yourself (if you have the skills), pay someone to fix it for you (if you can or must trade money for expediency), or wait for someone else with the skills and/or money necessary to fix it. Regards, - Brad *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 1:05 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.com wrote: My bad – “natively” means using the Queue command from the dialplan. Since the “powers that be” are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:57 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote: Do you use the Queue command “natively” or from the AGI? In the example you gave, if you did a “core show channels”, I assume that Agent007 would be idle, but ineligible for Queue activity. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501’s and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:11 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 10:36 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
snip Asterisk is (IMO) a very good product. It is NOT a perfect product, but I'm sure that most if not all of the Commercial PBX products available are not either. You get what you pay for; In this case, you pay in time instead of actual cash (unless you use the commercial flavor of Asterisk). It all boils down to what you need and what you are willing to do/pay to get that. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Actually from what I understand Asterisk is the only product that has this REFER problem. I know for a fact FreeSWITCH (open-source) handles REFERs fine. On Wed, Feb 23, 2011 at 1:28 PM, Danny Nicholas da...@debsinc.com wrote: snip Asterisk is (IMO) a very good product. It is NOT a perfect product, but I’m sure that most if not all of the Commercial PBX products available are not either. You get what you pay for; In this case, you pay in time instead of actual cash (unless you use the commercial flavor of Asterisk). It all boils down to what you need and what you are willing to do/pay to get that. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
You are still focusing on ONE of the choices given when that isn't your only option. It is simply untrue to say that the answer to it's broken was pay us. You were (now on multiple occasions) told how it would come to pass that a resolution will come about. You choose to ignore precisely two-thirds of the options available to you in order to continue to grind your axe. I am convinced you are either trolling or simply myopic. You have choices, they are yours to make. Stop trying to say that the entire Asterisk development community is simply in it for money, because that is patently false. - Brad From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) It's simple, if a product is broken shouldn't it be fixed? In this case the answer is for a price which is absurd because it is an open source product. If there was a decent community of developers surrounding this open source project, it would be fixed simply because it's broken, no questions asked. On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley bradley.watk...@compuware.commailto:bradley.watk...@compuware.com wrote: Implying that the Asterisk developers (which is itself a fairly nebulous statement since those who contribute to Asterisk are many and come from different companies/countries/etc.) are not in it to make a good product but to make a profit is not only highly insulting but a complete mischaracterization of what you were told on IRC. What you were told was that there are essentially three choices (and this goes for pretty much any open source software, as already stated). You may either fix it yourself (if you have the skills), pay someone to fix it for you (if you can or must trade money for expediency), or wait for someone else with the skills and/or money necessary to fix it. Regards, - Brad 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 vip killa Sent: Wednesday, February 23, 2011 1:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: My bad - natively means using the Queue command from the dialplan. Since the powers that be are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. 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 vip killa Sent: Wednesday, February 23, 2011 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: Do you use the Queue command natively or from the AGI? In the example you gave, if you did a core show channels, I assume that Agent007 would be idle, but ineligible for Queue activity. 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 vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.commailto:da...@debsinc.com wrote: I use Polycom 501's and
Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card
On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
This is the closest thing I was able to find in my wctdm.c file:
if ((blah 0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors when compling.
Maybe you need to remove the closing brace too?
--Greg
--
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? On Wed, Feb 23, 2011 at 1:38 PM, Watkins, Bradley bradley.watk...@compuware.com wrote: You are still focusing on ONE of the choices given when that isn’t your only option. It is simply untrue to say that the answer to “it’s broken” was “pay us”. You were (now on multiple occasions) told how it would come to pass that a resolution will come about. You choose to ignore precisely two-thirds of the options available to you in order to continue to grind your axe. I am convinced you are either trolling or simply myopic. You have choices, they are yours to make. Stop trying to say that the entire Asterisk development community is simply in it for money, because that is patently false. - Brad *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 1:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) It's simple, if a product is broken shouldn't it be fixed? In this case the answer is for a price which is absurd because it is an open source product. If there was a decent community of developers surrounding this open source project, it would be fixed simply because it's broken, no questions asked. On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley bradley.watk...@compuware.com wrote: Implying that the Asterisk developers (which is itself a fairly nebulous statement since those who contribute to Asterisk are many and come from different companies/countries/etc.) are “not in it to make a good product” but to make a “profit” is not only highly insulting but a complete mischaracterization of what you were told on IRC. What you were told was that there are essentially three choices (and this goes for pretty much any open source software, as already stated). You may either fix it yourself (if you have the skills), pay someone to fix it for you (if you can or must trade money for expediency), or wait for someone else with the skills and/or money necessary to fix it. Regards, - Brad *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 1:05 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Yes, they want money, they've told me that several times...it's unfortunate that asterisk's dev community is not in it to make a good product but a profit On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas da...@debsinc.com wrote: My bad – “natively” means using the Queue command from the dialplan. Since the “powers that be” are aware of this problem, I suppose it will get fixed when somebody either has some spare time or a sufficient bounty is offered. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:57 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote: Do you use the Queue command “natively” or from the AGI? In the example you gave, if you did a “core show channels”, I assume that Agent007 would be idle, but ineligible for Queue activity. -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa *Sent:* Wednesday, February 23, 2011 11:37 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential
Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
Sorry for the top post - this is from my phone. Sounds like the issue may actually be with the AGI that is handling your ACD queue. I've used the built-in Queue() command to handle situations like you describe without running into the issues you detailed. And that's with Polycom phones, too. Without more details, I'm not sure how much help you're going to get. Show us some console output of the issue, capture the proper debug logs, etc, and perhaps you'll find more help. Thanks, --Warren Selby, dCAP On Feb 23, 2011, at 11:57 AM, vip killa vipki...@gmail.com wrote: I'm sorry i don't know what you mean by natively. I'm almost certain the queue is handled via AGI and not using asterisk's queue. On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas da...@debsinc.com wrote: Do you use the Queue command “natively” or from the AGI? In the example you gave, if you did a “core show channels”, I assume that Agent007 would be idle, but ineligible for Queue activity. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Sure, it really manifests itself whenever using AGI for call flow, but this is how it affects us... incoming call - queue - agent007 - xfer - pussygalore now the AGI/dialplan thinks agent007 is on phone with pussygalore until that xfered call terminates so if another call comes into queue while pussygalore is on the phone w/ that xfered call, agent007 will not even be attempted by queue I'm sure there could be other scenarios in which this REFER issue could pose a problem but this is the most consequential scenario which we have to deal with everyday. On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas da...@debsinc.com wrote: I use Polycom 501’s and use the Transfer Key to send inbound calls to other extensions. Can you give me an A-B-C example of how this problem manifests itself? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) Interesting but the issue I'm having relates to Inbound and Outbound REFERs since I'm using Polycom's Transfer softkey (which allows for both Inbound and Outbound Transfers). I know this is not an issue when using Asterisk's built-in transfer (only allows Inbound transfers). On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas da...@debsinc.com wrote: Have you read this thread? http://forums.digium.com/viewtopic.php?t=74418 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I did not see this issue anywhere on issues.asterisk.org Can you give me a reference number to the issue? Also, it is a problem with all releases of asterisk. On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails. Thanks. I’m quite certain that this problem is being considered (for reference, this is a 1.8.X issue). If you aren’t satisfied with the progress being made, you should research your own solution and/or offer a bounty. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On 02/23/2011 12:43 PM, vip killa wrote: I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? Option 3 was wait for someone else with the skills and/or money necessary to fix it. Demanding that somebody fix an issue will not work in any community, open source or otherwise. You'll only be labeled a nuisance and ignored. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? snip From what I see, the source fix on the Asterisk level would indeed be a major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from Agent007 to pussygalore, Agent007 is marked as busy in the queue although the call is no longer active for 007. One possible workaround would be to have a duplicate bail queue set up the same way. If my AGI does a core show channels and sees that 007 is not on the phone, I can do queue(bail) instead of queue(normal). -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? In a commercial product, you have two options when you find a bug: (1) Pay for it to be fixed. (2) Live with the bug. In an open-sourced product, you have those same two options, plus an additional one: (3) Fix it yourself. Those are the only three you have to choose from. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, 23 Feb 2011, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951) I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? snip From what I see, the “source fix” on the Asterisk level would indeed be a major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let’s say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from Agent007 to pussygalore, Agent007 is marked as busy in the queue although the call is no longer active for 007. One possible workaround would be to have a duplicate “bail queue” set up the same way. If my AGI does a “core show channels” and sees that 007 is not on the phone, I can do queue(bail) instead of queue(normal). Watch out for race conditions doing things like this... j-- _ -- 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] SIP friend name
Is there a way to configure a friend in sip.conf that allows a station to register using a username other than the [name]? I want to have something like this in sip.conf: [1234] username=something_really_long_and_random secret=something_else_really_long_and_random ... Then allow a SIP REGISTER like so: REGISTER sip:10.0.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.201;branch=z9hG4bK9238ea1821361589 From: Joe User sip:something_really_long_and_random@10.0.0.200;tag=CA8D62BF-9750393E To: sip:something_really_long_and_random@10.0.0.200 CSeq: 2 REGISTER Call-ID: fbc4bb4b-91001d55-8c6da4c4@10.0.0.201 Contact: sip:something_really_long_and_random@10.0.0.200;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017 Accept-Language: en Authorization: Digest username=something_really_long_and_random, realm=asterisk, nonce=35d57376, uri=sip:10.0.0.201:5060, response=5a0596db3c3f1823b783ae195074cc5c, algorithm=MD5 Max-Forwards: 70 Expires: 3600 Content-Length: 0 And end up being able to reference the device via SIP/1234. I've tried adding name, username, auth, authname, defaultname, and fromname among others and none seem to do what I'm looking for. Tried reading through chan_sip.c but have not figured it out yet. Figured I'd ask. Thanks in advance, Paul PS: I had a SIP entry with too easy a secret get exploited to the tune of $500 worth of calls to Liberia... -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. On Wed, Feb 23, 2011 at 2:21 PM, Richard Kenner ken...@gnat.com wrote: I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? In a commercial product, you have two options when you find a bug: (1) Pay for it to be fixed. (2) Live with the bug. In an open-sourced product, you have those same two options, plus an additional one: (3) Fix it yourself. Those are the only three you have to choose from. -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, 23 Feb 2011, vip killa wrote: I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. You 'effing' kill me :) You have to be a troll. You can't be this stupid. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
On Wed, Feb 23, 2011 at 3:43 PM, vip killa vipki...@gmail.com wrote: I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. If you're really interested in trying to resolve your issue, as opposed to just complaining about it, perhaps you can post the requested debug information[1] from earlier. [1] - https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information With the requested information, you may be surprised at the type and level of help you may get from this mailing list. If all else fails, you can always open a new issue on the bug tracker and it will get looked at. It's a pretty painless procedure. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951) I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. It is free if you can use it. You can pay for all the help you want to or have the money to pay for. The Asterisk Software Charity Society went bankrupt about 2500 years ago. You can pick some name from the mail list and demand they fix the issue you perceive. But you probably won't be able to wait for results. There are 3 legs to any transaction. Speed, Quality, Price. You get to pick any two. The other party gets to set the 3rd one. You can't set all 3. 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] REFER and dialplan broken (asdocumentedinchan_sip.c on line 11951)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Wednesday, February 23, 2011 4:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] REFER and dialplan broken (asdocumentedinchan_sip.c on line 11951) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951) I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. It is free if you can use it. You can pay for all the help you want to or have the money to pay for. The Asterisk Software Charity Society went bankrupt about 2500 years ago. You can pick some name from the mail list and demand they fix the issue you perceive. But you probably won't be able to wait for results. There are 3 legs to any transaction. Speed, Quality, Price. You get to pick any two. The other party gets to set the 3rd one. You can't set all 3. Cary Except in Wisconsin. -- _ -- 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] REFER and dialplan broken(asdocumentedinchan_sip.c on line 11951)
It is free if you can use it. You can pay for all the help you want to or have the money to pay for. The Asterisk Software Charity Society went bankrupt about 2500 years ago. You can pick some name from the mail list and demand they fix the issue you perceive. But you probably won't be able to wait for results. There are 3 legs to any transaction. Speed, Quality, Price. You get to pick any two. The other party gets to set the 3rd one. You can't set all 3. Cary Except in Wisconsin. Even there. -- _ -- 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] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
On 11-02-23 10:31 AM, Jose P. Espinal wrote: - Added a new configuration option remotesecret for authentication to remote services. For backwards compatibility, secret still has the same function as before, but now you can configure both a remote secret and a local secret for mutual authentication. - I thought that 'remotesecret' is used to authenticate myself when placing a call to the remote network, as I used to do with 'secret' parameter. I may be mistaken, because I don't use remotesecret, but I think the purpose of that was to allow different authentication depending on the direction. My guess is remotesecret is used to authenticate the remote end when a call is placed into Asterisk, and secret is used when you're placing a call to the remote server. Or it's possible the feature has a bug and an issue should probably be opened on the issue tracker ;) Leif. -- _ -- 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] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
On 11-02-23 10:31 AM, Jose P. Espinal wrote: Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Actually I was wrong! See here. It is being resolved. https://reviewboard.asterisk.org/r/1107/ Leif. -- _ -- 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] One way dialing over a SIP trunk
I have a SIP trunk built between a Cisco CallManager version 8. I can dial the
phones registered to the Asterisk PBX from a phone registered to the Call
Manager.
I've tried to keep the config as small as possible to help the troubleshooting
process. Attached is he most recent debug.
My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251
SIP.CONF
[6001]
type=friend
secret=cisco2003
callerid=Dave 6001
host=dynamic
canreinvite=no
context=myphones
regexten=6001
[CM8]
type=friend
host=10.169.169.250
canreinvite=yes
;disallow=all
allow=ulaw
allow=alaw
qualify=yes
context=myphones
Extensions.conf
myphones]
exten = 6001,1,Dial(SIP/6001)
exten = 6001,2,Hangup()
exten = _X.,1,Dial(SIP/CM8/${EXTEN:0},30,rt)
Thanks for any help.
Mitch
{\rtf1\ansi\ansicpg1252\cocoartf1038\cocoasubrtf350
{\fonttbl\f0\fswiss\fcharset0 Helvetica;}
{\colortbl;\red255\green255\blue255;\red20\green54\blue165;}
\margl1440\margr1440\vieww12000\viewh8400\viewkind0
\deftab720
\pard\pardeftab720\ql\qnatural
\f0\fs24 \cf0 tunafish*CLI \
\
--- SIP read from UDP:{\field{\*\fldinst{HYPERLINK http://10.169.169.138:2048/}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} ---\
INVITE {\field{\*\fldinst{HYPERLINK mailto:sip%3A6500@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone SIP/2.0\
Via: SIP/2.0/UDP 10.169.169.138:2048;branch=z9hG4bK-k3a2um4e9a01;rport\
From: test 6002 {\field{\*\fldinst{HYPERLINK mailto:sip%3A6002@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.251}};tag=zv4a0j6k6y\
To: {\field{\*\fldinst{HYPERLINK mailto:sip%3A6500@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone\
Call-ID: 3c26ed7edf41-m2sk5pf5ralb\
CSeq: 1 INVITE\
Max-Forwards: 70\
Contact: {\field{\*\fldinst{HYPERLINK sip:6002@10.169.169.138:2048;line=vtmynxjj}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.138:2048;line=vtmynxjj}};reg-id=1\
X-Serialnumber: 000413347AE4\
P-Key-Flags: keys=3\
User-Agent: snom300/8.4.18\
Accept: application/sdp\
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE\
Allow-Events: talk, hold, refer, call-info\
Supported: timer, 100rel, replaces, from-change\
Session-Expires: 3600;refresher=uas\
Min-SE: 90\
Content-Type: application/sdp\
Content-Length: 395\
\
v=0\
o=root 1038861251 1038861251 IN IP4 10.169.169.138\
s=call\
c=IN IP4 10.169.169.138\
t=0 0\
m=audio 59248 RTP/AVP 0 8 9 99 3 18 4 101\
a=rtpmap:0 pcmu/8000\
a=rtpmap:8 pcma/8000\
a=rtpmap:9 g722/8000\
a=rtpmap:99 g726-32/8000\
a=rtpmap:3 gsm/8000\
a=rtpmap:18 g729/8000\
a=fmtp:18 annexb=no\
a=rtpmap:4 g723/8000\
a=rtpmap:101 telephone-event/8000\
a=fmtp:101 0-16\
a=ptime:20\
a=sendrecv\
-\
--- (19 headers 18 lines) ---\
Sending to {\field{\*\fldinst{HYPERLINK http://10.169.169.138:2048/}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} (no NAT)\
Using INVITE request as basis request - 3c26ed7edf41-m2sk5pf5ralb\
Found peer '6002' for '6002' from {\field{\*\fldinst{HYPERLINK http://10.169.169.138:2048/}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}}\
\
--- Reliably Transmitting (no NAT) to {\field{\*\fldinst{HYPERLINK http://10.169.169.138:2048/}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} ---\
SIP/2.0 401 Unauthorized\
Via: SIP/2.0/UDP 10.169.169.138:2048;branch=z9hG4bK-k3a2um4e9a01;received=10.169.169.138;rport=2048\
From: test 6002 {\field{\*\fldinst{HYPERLINK mailto:sip%3A6002@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.251}};tag=zv4a0j6k6y\
To: {\field{\*\fldinst{HYPERLINK mailto:sip%3A6500@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone;tag=as2244e25f\
Call-ID: 3c26ed7edf41-m2sk5pf5ralb\
CSeq: 1 INVITE\
Server: Asterisk PBX 1.8.2.4\
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\
Supported: replaces, timer\
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7a1c370b\
Content-Length: 0\
\
\
\
Scheduling destruction of SIP dialog '3c26ed7edf41-m2sk5pf5ralb' in 32000 ms (Method: INVITE)\
\
--- SIP read from UDP:{\field{\*\fldinst{HYPERLINK http://10.169.169.138:2048/}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} ---\
ACK {\field{\*\fldinst{HYPERLINK mailto:sip%3A6500@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone SIP/2.0\
Via: SIP/2.0/UDP 10.169.169.138:2048;branch=z9hG4bK-k3a2um4e9a01;rport\
From: test 6002 {\field{\*\fldinst{HYPERLINK mailto:sip%3A6002@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.251}};tag=zv4a0j6k6y\
To: {\field{\*\fldinst{HYPERLINK mailto:sip%3A6500@10.169.169.251}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone;tag=as2244e25f\
Call-ID: 3c26ed7edf41-m2sk5pf5ralb\
CSeq: 1 ACK\
Max-Forwards: 70\
Contact: {\field{\*\fldinst{HYPERLINK sip:6002@10.169.169.138:2048;line=vtmynxjj}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.138:2048;line=vtmynxjj}};reg-id=1\
Content-Length: 0\
\
-\
--- (9 headers 0 lines) ---\
\
--- SIP read from
Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
On 02/23/2011 08:56 PM, Leif Madsen wrote: Actually I was wrong! See here. It is being resolved. https://reviewboard.asterisk.org/r/1107/ Leif. Thanks for the feedback, Leif! I will follow that incident closely, as I was starting to doubt about my understanding of English (jk) -- Jose P. Espinal http://www.eslackware.com IRC: [OFTC|FreeNode] Khratos @ #slackware | #asterisk/-doc/-bugs -- _ -- 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] Problem installing FXS module in old digium 4 channel tdm card
This worked.
Thank you all for your help.
On Wed, Feb 23, 2011 at 1:42 PM, Greg Woods g...@gregandeva.net wrote:
On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
This is the closest thing I was able to find in my wctdm.c file:
if ((blah 0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc-flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors when compling.
Maybe you need to remove the closing brace too?
--Greg
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Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1
On Feb 23, 2011, at 7:11 PM, Jose P. Espinal wrote: On 02/23/2011 08:56 PM, Leif Madsen wrote: Actually I was wrong! See here. It is being resolved. https://reviewboard.asterisk.org/r/1107/ Leif. Thanks for the feedback, Leif! I will follow that incident closely, as I was starting to doubt about my understanding of English (jk) I had forgotten that I got a Ship It! on that patch. I went ahead and committed the fix to 1.6.2, 1.8, and trunk. Terry -- _ -- 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] alarm POTS lines
On Thu, Dec 2, 2010 at 11:58, Jeff LaCoursiere j...@sunfone.com wrote: we have a low-cost Atom based PBX and a fax relay setup locally with hylafax/iaxmodem to solve that issue, and it is working very well. We don't however, have a solution for their alarm lines. You would desire the entire path to be UL listed if you are doing anything other than facilitating the phone call to the central station. There is app_alarmreciever in Asterisk, and furthermore the ContactID protocol is pure DTMF so that should work without issues. But why use phone lines at all? Recently I installed a DSC T-Link TL260GS which uses internet and GSM, there is no phone line plugged into the alarm panel at all. The problem is of course that modem calls over VoIP are flaky at best. Even though these alarm calls are low baud rate, when we test with the alarm company we only pass about 30% of the time (ulaw from customer site to our central switch, then out a T1). To be fair there is no QoS on their Internet links yet, and that certainly plays a role. SIA format is 110 or 300 baud, ContactID is (rapid) DTMF. -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- 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] DIAL through Specific number in PRI
Hi ALL, I have PRI line everything is fine , but my customer having a requirement that they want to DIAL a number from PRI which gives callerid as Specific number. i.e PRI start from 3055 to 30550100 i have purchased a 100 number from telco and our pilot number is 3055, now if some caller want to dial any number but caller should shown is 30550008 like this. is there any solution from asterisk side. regards Dhaval -- _ -- 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] DIAL through Specific number in PRI
If your PRI provider permit you to associate any ANI to any Circuit-ID you can do this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, February 24, 2011 12:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DIAL through Specific number in PRI Hi ALL, I have PRI line everything is fine , but my customer having a requirement that they want to DIAL a number from PRI which gives callerid as Specific number. i.e PRI start from 3055 to 30550100 i have purchased a 100 number from telco and our pilot number is 3055, now if some caller want to dial any number but caller should shown is 30550008 like this. is there any solution from asterisk side. regards Dhaval -- _ -- 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] AMI FullyBooted issue
Hi, I have this same behaviour on version 1.8.2.3 build from source. We are using AMI to originate call from our CRM software, but we ignore that message. Regards, Marcin -- _ -- 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
