Re: [asterisk-users] problems with natted phones
El jue, 8 de jul. de 2021 a la(s) 14:58, Marek Greško (mgres...@gmail.com) escribió: > The asterisk is connected to the internet with public static IP address. > > The pjsip config contains: > > What does your transport config look like? Take a look at this wiki page: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT -- Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink
- On Feb 5, 2021, at 11:18 AM, Michael L. Young wrote: > - On Feb 4, 2021, at 4:26 PM, Social Boh wrote: >> The problem is with this CentOS 7 glibc version: >> 2.17-317.el7 >> After the library update and system reboog, >> gotoif Asterisk application, stop to working >> Any hint to solve? > Until it is resolved, you can do a 'yum history' and note the transaction ID > of > the update. Then try running 'yum history undo [transaction id]'. That should > roll you back to the previous glibc. > Looks like Red Hat is already working on it: > https://access.redhat.com/solutions/5778071 Here is the Bugzilla report for anyone on RHEL / CentOS 7: https://bugzilla.redhat.com/show_bug.cgi?id=1925204 -- Michael Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CentOS 7 yum-update and Gotoif has stoppped to workink
- On Feb 4, 2021, at 4:26 PM, Social Boh wrote: > The problem is with this CentOS 7 glibc version: > 2.17-317.el7 > After the library update and system reboog, > gotoif Asterisk application, stop to working > Any hint to solve? Until it is resolved, you can do a 'yum history' and note the transaction ID of the update. Then try running 'yum history undo [transaction id]'. That should roll you back to the previous glibc. Looks like Red Hat is already working on it: https://access.redhat.com/solutions/5778071 -- Michael Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?
> From: "John Hughes" > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > > Sent: Thursday, May 14, 2020 2:10:45 AM > Subject: [asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and > alaw; asterisk wants to translate g729 -> alaw. WHY? > I am having a problem with one of my callers who is using either g729 or > alaw. I > can do alaw but not g729 so asterisk should negotiate alaw right? In fact from > the sip debug it looks like it does, but then I get the dreaded > "channel.c:5630 > set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the > call hangs up. Why? > Last minute thought: Is it possible that the caller is sending g729 in RTP > even > though the SIP negotiation clearly chooses alaw? Maybe I need some RTP > debugging. > Asterisk 13.14.1 on Debian, using chan_sip. Hi John, Maybe a newer version of Asterisk would help? The latest release for 13 is version 13.33. The version you are on was released 3 years ago. Here is an issue which looks like what you describe and was fixed in 13.16 [ https://issues.asterisk.org/jira/browse/ASTERISK-26143 | https://issues.asterisk.org/jira/browse/ASTERISK-26143 ] Not sure if this is the answer to your problem but thought that I would throw that out there. Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] permission woes on systemd
- Original Message - > From: "sean darcy" > To: "Asterisk Users Mailing List, Non-Commercial Discussion" > > Sent: Tuesday, January 21, 2020 9:22:28 PM > Subject: [asterisk-users] permission woes on systemd [..] > So why would starting asterisk as user asterisk work, but fail using > systemd ? > Have you checked SELinux? After creating the configuration files, did you run 'restorecon' on the appropriate asterisk directories? If not, the files are not labeled correctly and SELinux might be denying access. Just a thought. Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 100% CPU after upgrade.
- Original Message - > From: "Mike Diehl" > Sent: Monday, April 3, 2017 5:45:58 PM > Subject: Re: [asterisk-users] 100% CPU after upgrade. > Those are all rational questions, so here we go: > > We upgraded from 11.x, though the system was a backup server, so it was never > actually used. > > The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should have plenty > of power for what I'm asking it to do. The system is configured via RT using > a local Mysql database. > Which distro are you running? How are you starting Asterisk (init script / systemd)? -- Michael L. Young -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibilty between agi for asterisk 13.8.0 and php5.6
- On May 4, 2016, at 8:49 AM, Mamadou NGOM n...@numericap.com wrote: > Hello everybody, > When I call my extension the agi script don't work well. when I look at the > cli, > that is what I have: > AGI Tx >> agi_request: **.php > AGI Tx >> agi_channel: SIP/myprovider-0007 > AGI Tx >> agi_language: fr > AGI Tx >> agi_type: SIP > AGI Tx >> agi_uniqueid: *** > AGI Tx >> agi_version: 13.8.0 > AGI Tx >> agi_callerid:* > AGI Tx >> agi_calleridname: unknown > AGI Tx >> agi_callingpres: 0 > AGI Tx >> agi_callingani2: 0 > AGI Tx >> agi_callington: 0 > AGI Tx >> agi_callingtns: 0 > AGI Tx >> agi_dnid: > AGI Tx >> agi_rdnis: unknown > AGI Tx >> agi_context: default > AGI Tx >> agi_extension: > AGI Tx >> agi_priority: 13 > AGI Tx >> agi_enhanced: 0.0 > AGI Tx >> agi_accountcode: > AGI Tx >> agi_threadid: * > AGI Tx >> agi_arg_1: 56 > AGI Tx >> > AGI Rx << SET VARIABLE ** 2 > AGI Tx >> 510 Invalid or unknown command > -- AGI Script ***.php completed, returning 0 > I looked on the Internet but I saw a clear answer > it is sure that it is for the compatibility between php5.6 and agi. if > somebody > can help me. Make sure there are no windows or dos line endings in that php script. Try running it through dos2unix and see if that solves your issue. Regards, Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.10.0 Now Available
- Original Message - > From: cov...@ccs.covici.com > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Thursday, May 29, 2014 6:42:05 PM > Subject: Re: [asterisk-users] Asterisk 11.10.0 Now Available > > > * ASTERISK-23754 - [patch] Use var/lib directory for log file > > configured in asterisk.conf (Reported by Igor Goncharovsky) > Is this mandatory -- what is wrong with /var/log/asterisk for those > files? > The title on that issue is very misleading. The patch that went in was just for chan_ooh323. The change was to have chan_ooh323 use the log directory configured in asterisk.conf instead of using a hard coded value. Michael -- _ -- 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] Login by AMI ok, by AJAM fails
- Original Message - > From: "Michael L. Young" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, May 16, 2014 4:55:30 PM > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > - Original Message - > > From: "Michelle Dupuis" > > To: "Asterisk Users List" > > Sent: Friday, May 16, 2014 4:29:05 PM > > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > > > From: asterisk-users-boun...@lists.digium.com > > on behalf of Michael L. > > Young > > Sent: Friday, May 16, 2014 4:16 PM > > To: Asterisk Users List > > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > > > Have you taken a look at the Wiki yet? > > > > https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP > > > > In looking at that, I see some mistakes in what you are trying to > > do. > > Please take a look at that and give it a try. > > Well, you need to login first. Since you are using cURL, you need to > turn the cookie engine on so that it will store and send cookies. > > Also, you need to send the login request to > "http://localhost:5039/asterisk/manager?action=login&user=test&secret=test"; > and not "rawman". Once you are logged in, then you can get raw > manager output. > > I hope that helps. > > Michael Sorry that I messed up the thread while trying to un-top post your message. The above was in response to your prior message: > > I've done all of that (and I set the AJAM to listen to 5039). What > > mistakes do you see? Michael -- _ -- 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] Login by AMI ok, by AJAM fails
- Original Message - > From: "Michelle Dupuis" > To: "Asterisk Users List" > Sent: Friday, May 16, 2014 4:29:05 PM > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > From: asterisk-users-boun...@lists.digium.com > on behalf of Michael L. > Young > Sent: Friday, May 16, 2014 4:16 PM > To: Asterisk Users List > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > Have you taken a look at the Wiki yet? > > https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP > > In looking at that, I see some mistakes in what you are trying to do. > Please take a look at that and give it a try. Well, you need to login first. Since you are using cURL, you need to turn the cookie engine on so that it will store and send cookies. Also, you need to send the login request to "http://localhost:5039/asterisk/manager?action=login&user=test&secret=test"; and not "rawman". Once you are logged in, then you can get raw manager output. I hope that helps. Michael -- _ -- 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] Login by AMI ok, by AJAM fails
- Original Message - > From: "Michelle Dupuis" > To: "Asterisk Users List" > Sent: Friday, May 16, 2014 3:39:35 PM > Subject: Re: [asterisk-users] Login by AMI ok, by AJAM fails > > You're right - but I tried username too and it fails. I can't > understand why AMI authenticates and AJAM fails... > Have you taken a look at the Wiki yet? https://wiki.asterisk.org/wiki/display/AST/Allow+Manager+Access+via+HTTP In looking at that, I see some mistakes in what you are trying to do. Please take a look at that and give it a try. Michael -- _ -- 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] Login by AMI ok, by AJAM fails
- Original Message - > From: "Michelle Dupuis" > To: "Asterisk Users List" > Sent: Friday, May 16, 2014 2:43:30 PM > Subject: [asterisk-users] Login by AMI ok, by AJAM fails > -- > root@apbx:/tmp# curl > http://localhost:5039/asterisk/rawman?action=login&user=test&secret=test > [1] 15548 > [2] 15549 > root@pbx:/tmp# Response: Error > Message: Authentication failed > [1]- Done curl http://localhost:5039/asterisk/rawman?action=login > [2]+ Done user=test I believe it should be "username" instead of "user" for the query parameter. Michael -- _ -- 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] Security log format / content
- Original Message - > From: "Michelle Dupuis" > To: "Asterisk Users List" > Sent: Thursday, March 27, 2014 12:55:21 AM > Subject: [asterisk-users] Security log format / content > I've noticed that the Asterisk (v11) security log captures attempts > do dial without first authenticating, and places the number dialed > into the "accountid" field. > I'm trying to distinguish between failed attempts to register and > attempts to dial without registering, but the security log treats > them identically (using the accountid field for either the username > or number dialed). I have noticed that the eventversion field is set > to 2 for failed dial attempts, and 1 otherwise. > Is this coincidence? Or can I rely on the eventversion=2 in the > future to distinguish these two event types? (I've looked here: > https://wiki.asterisk.org/wiki/display/AST/Security+Log+File+Format > but it doesn't really help) The "eventversion" field is just a way to distinguish different versions of the same event. Between Asterisk 10 and 11, that particular event's logging output changed requiring a bump up in the version. It should not be used to distinguish different events. What do you mean by "eventversion field is set to 2 for failed dial attempts, and 1 otherwise"? What is the event? I have a feeling those are two different events. You are correct about the events looking identical whether it is a failed registration or a failed dial attempt. From the standpoint of Asterisk, an attempt was made to either register or place a call but the credentials failed. Therefore, an "InvalidPassword" event is logged. When an authorized device successfully places a call, you will only have a "ChallengeSent" entry in your log. If an attempt to place a call is made and it does not respond back with the right credentials to the challenge sent to Asterisk, then you will have a "ChallengeSent" entry with a subsequent "InvalidPassword". You should be able to connect the two events based on the fields in those events. If a successful attempt to register is made, you will have a "ChallengeSent" with a subsequent "SuccessfulAuth". If it is not successful, then you will have a "ChallengeSent" with a subsequent "InvalidPassword". Again, there should be enough information present with the other fields to help connect the events together. The security events in Asterisk are designed to present the events. It does not determine anything else for you. You have to create a consumer of those events that can attempt to connect the dots for you. Hopefully we are providing enough information for the consumer to do whatever you would like the consumer to do with the information. I hope that helps. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ignoring nat settings
- Original Message - > From: "Andres" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Thursday, January 16, 2014 4:17:53 PM > Subject: Re: [asterisk-users] Asterisk ignoring nat settings > > > I am curious why you would say that "nat=yes" might work over > > "nat=force_rport,comedia"? As you stated, they are the same. > > "nat=yes" is deprecated and should be discouraged from being > > used. > I had no idea it was deprecated. I have never seen such a warning in > Asterisk 1.8.X The OP didn't specify which version of Asterisk he was using. In Asterisk 1.8, "nat" was not a combinable list of options. In Asterisk 11 it was. So, I figured that since he was asking about "nat=force_rport,comedia" that he was on Asterisk 11 and in that version, "nat=yes" is deprecated. I apologize about not clarifying the version that I was talking about. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk ignoring nat settings
- Original Message - > From: "Andres" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, January 15, 2014 7:51:28 PM > Subject: Re: [asterisk-users] Asterisk ignoring nat settings > Why don't you try with nat=yes. It should be equivalent to what you > have but who knows. It might just work. I am curious why you would say that "nat=yes" might work over "nat=force_rport,comedia"? As you stated, they are the same. "nat=yes" is deprecated and should be discouraged from being used. Michael -- _ -- 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] 11.5.0 - SIP registration not retrying after failures
> From: "Tony Mountifield" > To: asterisk-users@lists.digium.com > Sent: Friday, November 8, 2013 10:39:25 AM > Subject: [asterisk-users] 11.5.0 - SIP registration not retrying after > failures > > I had a SIP problem on an 11.5.0 system that I look after. It > registers > with a SIP trunk provider, and at one point the provider had an issue > that > caused registration to fail. > > The problem was that Asterisk did not keep retrying, and it was not > until > it was restarted that registration was re-established. A fix for this was actually just committed and will be in 11.7. There is a release candidate available if you want to try it out. You want to look for the "register_retry_403" option that was added to the sip.conf file. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.5 not honoring RTP port change in RE-INVITE
- Original Message - > From: "Noah Engelberth" > I have an Asterisk 11.5 system, using SIP Realtime and operating as a > ITSP. One of my customer’s endpoints is a NetVanta 7100 PBX system > that has a SIP trunk connection to my Asterisk box. The NV 7100 has > a public IP on it that doesn’t have any NAT between it and my > Asterisk system. When the customer transfers a call from one handset > to a voicemail box, the NV 7100 sends a RE-INVITE to Asterisk with > SDP information for a different RTP port number. Asterisk is ACKing > the RE-INVITE, but never changes media over to the new port number. > AdTran is saying it’s Asterisk’s problem, since the Wireshark trace > shows Asterisk is ACKing the re-invite but not changing ports. I do > see that the Session ID number is different in the two invites (the > REINVITE has a higher ID number than the original 200 OK that sets > up the call – my test call was inbound to the NV7100). However, the > REINVITE’s version number is lower (1) than the 200 OK’s SDP version > number (which was the same as the SDP Session ID number). I see in > the sip.conf.sample file that “By default, Asterisk will honor the > session version number in SDP packets and will only modify the SDP > session if the version number changes”. Given that I don’t have > ignoresdpversion=yes either globally or for this peer, does this > mean that Asterisk will only honor new SDP packets if the version is > higher, or will it honor any change? Or should I be looking > somewhere else? You have pretty much found what the issue is. The AdTran is not properly incrementing the SDP version. Look at the comments on these issues for clarification on why Asterisk is actually following the RFC3264: https://issues.asterisk.org/jira/browse/ASTERISK-20633 https://issues.asterisk.org/jira/browse/ASTERISK-20642 https://issues.asterisk.org/jira/browse/ASTERISK-21411 RFC3264 "If the offered SDP is different from the previous SDP, some constraints are placed on its construction, discussed below." "Nearly all aspects of the session can be modified. New streams can be added, existing streams can be deleted, and parameters of existing streams can change. When issuing an offer that modifies the session, the "o=" line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP." Therefore, in order to work with devices that do not handle the SDP version properly, sip.conf has the "ignoresdpversion" option. Michael (elguero) -- _ -- 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] 811
Quoting Mike Diehl : Is there a list somewhere? There is a list by state here: http://www.call811.com/state-specific.aspx -- _ -- 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] External sip phones register with the servers IP...
- Original Message - > From: "Carlos Chavez" > To: asterisk-users@lists.digium.com > Sent: Thursday, August 1, 2013 8:41:19 PM > Subject: [asterisk-users] External sip phones register with the servers IP... > > We have just updated our office server to Asterisk 11.4.0 from 1.8.15 > and > internally everything is working fine. The problem we are having is > that we > cannot use any external phone connected through the Internet. This > used to > work fine with 1.8 but since the upgrade whenever you register any > phone from > an outside network the phone tries to register using the servers > internal IP. > > I endo up having something like this: > > Sending to 187.163.93.235:58545 (no NAT) > -- Registered SIP '2003' at 192.168.2.50:58545 > Reliably Transmitting (no NAT) to 192.168.2.50:58545: > OPTIONS sip:2003@192.168.2.50:58545;ob SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.50:5060;branch=z9hG4bK5f2019c0 > Max-Forwards: 70 > From: "asterisk" ;tag=as4ed13172 > To: > Contact: > Call-ID: 46fd0ef840d6781d219269ae415e156e@192.168.2.50:5060 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 11.4.0 > Date: Fri, 02 Aug 2013 00:27:48 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH > Supported: replaces, timer > Content-Length: 0 > > I really cannot understand what is wrong, I have checked my sip.conf > configuration and it is the same as in past versions. externaddr and > localnet > are set to the proper values. Any ideas? Did you look at the CHANGES file? There are new settings for NAT. If you are using the same settings as in 1.8, there is a posiblity that you will have problems depending on what settings you have (which you did not include in this message). Also, I would recommend 11.5 since there was a one-way audio issue fixed related to using the two new NAT settings. -- Michael (elguero) -- _ -- 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] Playing a sound file during a call
- Original Message - > From: "Richard Mudgett" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Thursday, May 2, 2013 8:24:49 PM > Subject: Re: [asterisk-users] Playing a sound file during a call > > > On Thu, May 2, 2013 at 3:37 PM, Carlos Alvarez > > > > wrote: > > > > In case anyone else sees this discussion in the future, the > > Set(__DYNAMIC_FEATURES) line can't be over a certain length or it > > stops parsing anything after that. > > > > You can also put dynamic feature group names into the > DYNAMIC_FEATURES list. Also, I know this doesn't help you now but, in Asterisk 12 the limit has been eliminated. Take a look at https://issues.asterisk.org/jira/browse/ASTERISK-20680 Michael-- _ -- 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] rtcachefriends and rtautoclear on change password
- Original Message - > From: "Leandro Dardini" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, March 26, 2013 5:28:22 AM > Subject: [asterisk-users] rtcachefriends and rtautoclear on change password > > Hello friends, > I am using from a long time rtcachefirends=yes and rtautoclear=yes in > my sip.conf for asterisk 11.2.1. > > I have found the data of the peers are never reloaded from the > database, so if you change the password for a peer, it will continue > to work with the old password. Do you think it is the expected > behaviour? > > From the documentation for rtautoclear=yes > > If set to yes, when the registration expires, the friend will > vanish from the configuration until requested again. If set > to an integer, friends expire within this number of seconds > instead of the registration interval. > > The phone will renew the registration before it expires, so maybe it > never "expires". > > I have tried to set the rtautoclear to 60, but the result is the > same, > the new password is never enforced. > > Any suggestion apart from removing the rtcachefriends? With rtcachefriends turned on, the realtime peer is cached in memory. Therefore, in order to clear the cache for that peer, you should check into issuing the command "sip prune realtime peer " if you want to clear out only the one peer. If you want to reload the peer back in memory after clearing it out, you can issue "sip show peer load" to load it back from the db. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and dual stack support
- Original Message - > From: "Jaap Winius" > To: asterisk-users@lists.digium.com > Sent: Thursday, March 21, 2013 5:27:37 PM > Subject: Re: [asterisk-users] Asterisk 1.8 and dual stack support > > That's what I thought would happen. When I set bindaddr=:: and use > 'netstat -lpn |grep 5060' it shows: > > udp6 0 0 :::5060 :::* 9898/asterisk > > Services like this usually also support IPv4 and as much is suggested > by > this comment in the sip.conf that comes with my Asterisk package: > > ; (Note that using bindaddr=:: will show only a single > ; IPv6 socket in netstat. IPv4 is supported at the same > ; time using IPv4-mapped IPv6 addresses.) > > However, the moment I reload my sip.conf with bindaddr=::, my entire > list > of IPv4-only peers loses contact with Asterisk with warnings about > the > network being unreachable. So, it would appear that the version of > Asterisk that I'm using is operating with a single stack socket. Let me try to understand this. With bindaddr set as "bindaddr=::", upon starting Asterisk, you are fine and all your IPv4 peers connect properly. Therefore, dual stack is working at this point. Upon issuing a "sip reload", your peers lose their ability to communicate with Asterisk? Is that correct? What does "netstat -lpn |grep 5060" show after the reload? These "network unreachable" warnings are from Asterisk or your peers? What version of Asterisk are you using? Asterisk 1.8.0 had IPv6 support in it. Therefore, every minor version released since would still have IPv6 support in it. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 and dual stack support
- Original Message - > From: "Jaap Winius" > To: asterisk-users@lists.digium.com > Sent: Thursday, March 21, 2013 12:47:57 PM > Subject: [asterisk-users] Asterisk 1.8 and dual stack support > > Hi folks, > > Following an upgrade to Debian wheezy, I'm now running Asterisk > 1.8.13.1. > As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version > can > support IPv6. However, it seems that I can't get it to support both > IPv4 > and IPv6 at the same time. For example, if in sip.conf I set the > bindaddr > variable to '::' it will only listen on IPv6 and none of my IPv4-only > friends and peers will be able to connect to it. On the other hand, > if I > set it to '0.0.0.0' then it will not listen on IPv6. How are you determining that it is not listening on IPv4? bindaddr=:: should allow you to support dual stack. Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 Streaming MOH timing interface
- Original Message - > From: "Bob Pierce" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Cc: g...@westmancom.com > Sent: Monday, February 4, 2013 6:14:26 PM > Subject: [asterisk-users] Asterisk 1.8 Streaming MOH timing interface > We are running Asterisk 1.8.5.0 with an uptime of 40 weeks. Just > today our streaming music on hold stopped working. I remember when > we had first installed 1.8 we had an issue where the streaming music > on hold would not work because Music On Hold was using the DAHDI > timing module. We needed the DAHDI timing module loaded so that > paging would work. However, at that time we upgraded to 1.8.5.0 and > the system loaded properly with both the dahdi and pthread timing > module with Music On Hold using the pthread timing module. In that > state, everything worked properly - Streaming Music On Hold worked > as well as Paging. That has all continued to work properly for the > last 40 weeks. > I'm wondering of for some reason the Music on Hold service is now > using the DAHDI timing module because when I do "module show like > timing" I see: > CLI> module show like timing > Module Description Use Count > res_timing_dahdi.so DAHDI Timing Interface 33 > res_timing_pthread.so pthread Timing Interface 0 > 2 modules loaded > I believe that the pthread used to show a use count of at least 1 > with the Music On Hold service using that timing source. I suspec > that if I restart the Asterisk service everything will come back up > the way that it did last time. However, I'm wondering if there would > be a way to switch the Music On Hold module back to using pthread > timing without restarting the Asterisk service. Bob, I would recommend upgrading to the latest version. There have been a lot of security and bug fixes since 1.8.5. There was a bug fixed, over a year ago (1.8.9), which sounds exactly like what you are experiencing. The latest version is 1.8.20.1. Regards, Michael (elguero) -- _ -- 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] Verizon SIP "trunking" Field Trial
- Original Message - > From: "Logan Bibby" > Does anyone have a good contact for their sales? I've attempted > calling their Enterprise sales a few times and was just spun around > in circles. Having a sales rep I can just call would be awesome. Logan, We have an account manager that we deal with directly for changes or new orders. Supposedly, every customer has their own account manager. Michael -- _ -- 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] Verizon SIP "trunking" Field Trial
- Original Message - > From: "Matthew J. Roth" > At least Verizon maintains a consistent customer experience. ; ) > > Overall, we've found the service to be reliable and stable, but when > there are problems or changes needed you're dealing with Verizon and > the > w...h...e...e...l...s..t...u...r...n..s...l...o...w...l...y. Haha... that is funny... it is sooo true. Well, you are right. Once it is working, it is usually pretty stable. Just a pain in the butt when things are not working. Hopefully we can get through the Field Trial and that is all I have to worry about for a while. Thanks Matthew for all the encouragement as I go down this temporary (I hope) unpleasant path. Michael -- _ -- 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] Verizon SIP "trunking" Field Trial
- Original Message -
> From: "Carlos Alvarez"
> Sounds like the same huge effort it takes to work with
> Qwest/Centurylink, and in the long run we found it simply isn't
> worth it. The few benefits of working with an RBOC are countered by
> the many drawbacks of working with an RBOC.
> Also we recently acquired a half million minutes/mo from a company
> who was tired of dealing with Qwest SIP. They said the same thing I
> said above.
> I suppose the point of what I'm saying is you should really think
> about what you think you will gain from a relationship with them,
> and whether all this is worth it ("all this" means now and how their
> attitude will affect you forever).
Trust me, this was not my choice... They are not fun to deal with when it came
to our PRI lines. After dealing with dropped calls and errors on the T1s, they
wouldn't admit that they had a problem until finally they looked at the
hardware at the CO while we were down hard (which cost us about 4 - 6 hours
downtime) and said, "Oh, we do have a problem". To make a long story short, it
was fixed and has been good since but I was really trying to move us away from
Verizon. Unfortanately, it boils down to cost and Verizon being as big as they
are were able to make a deal (getting us out of contracts that had been signed,
credits, etc.) that the ultimate decision maker here at the company went for.
That decision maker also has the mindset that we have to stick with the phone
company for some reason. I was strongly against it and wanted to go with a
different company. So, I have to deal with it now.
Thanks for your input. It pretty much echoes my sentiments.
Michael
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Re: [asterisk-users] Verizon SIP "trunking" Field Trial
- Original Message - > From: "Matthew J. Roth" > Your email documents the same experience we had years ago. It was > strange reading it and I was shocked that nothing has changed in that > much time. Asterisk will work with Verizon's IP trunking product, > but > they're trying to make you jump through some old hoops first. Those were my thoughts. They are making this a lot more complicated than it really needs to be. I think the main thing they are worried about is having to support something they don't know anything about. Well, they won't have to support it; we will. Just as long as they are SIP compliant, there should be no issues. > We were using Verizon IP trunks over an MPLS network in 2008. At the > time, they did not require IPSEC for signaling. However, they did > want us to install an SBC and actually provided us with an AudioCodes > nCite 1000 at their cost. It just acted as a proxy, so it didn't > affect interoperability with Verizon's IP trunks and I wouldn't > buy one only to satisfy them. One of the engineers stated that they have received the direction to only use "standard" equipment. So, they are afraid that our setup will not pass ICB since it doesn't fit into their "standard" way of doing things. > We were quite happy with the service, so I'd encourage you to go > ahead > with the field trial without putting an SBC in place. Remember that > you will be paying them, so they should be working to fit your design > and if they reject you for some arcane reason then you are better off > with another provider anyway. > > Don't hesitate to let them know that you know you're jumping through > the same hoops that have been in place since 2008 and you'd > appreciate > it if they would streamline the process to save time and money. Tell > them that Asterisk should already be on their certified list of > approved devices because they've been running field trials and > production setups on it for years. It is good to hear that you were happy with the service. I have my reservations with all the hoops they are making us jump through and that gives me a bit of confidence that it will be worth it. I did tell them that Asterisk is being used all over the place as well as in big call centers. I know that I have seen others in the Asterisk community on Verizon. Verizon seems to be hung up on this certification stuff and it is hard to explain to them that this is not a piece of hardware you buy and plugin. We can build our own servers and put Asterisk on it, and they seem to cringe when they hear that. Thanks for your input Matthew. It is appreciated. Michael -- _ -- 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] Verizon SIP "trunking" Field Trial
> From: "Carlos Alvarez" > It may be too late for this, but in working with another RBOC who > didn't want to deal with Asterisk, I just asked what they do > support, and modified the headers sent by Asterisk to claim that it > was one of the devices on that list. Done. Like everyone else, I was laughing as well when I read this. One engineer stated that they like to have an SBC to manipulate the headers to normalize things. I stated that Asterisk was capable of manipulating headers if need be. You just proved that it works :) Michael -- _ -- 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] Verizon SIP "trunking" Field Trial
- Original Message - > From: "Steven Howes" > > I *think* Verizon require IPSEC for the signalling, so it may be > worth reading up on configuring IPSEC in Linux (or acquiring > additional hardware) whilst you're looking at the Asterisk part. > This could have just been for a specific product / contract or > something, I don't recall the details exactly. I should have probably stated that this is going to be going through an MPLS network being setup with Verizon. They may not be requiring that since it is within their network, not going over the internet. They have not said anything about the the need to secure the traffic coming from them or to them since the VoIP traffic will be on Verizon's network. Thanks for the heads up, though. I will keep that in mind. Michael -- _ -- 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] Verizon SIP "trunking" Field Trial
All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We have used other providers without any issues with our current setup but it seems that Verizon has their own standards when it comes to this and they don't seem very keen on linux and open source. Yet, they are willing to work with us and want to see the field trial succeed instead of being rejected from another group within Verizon who will have to approve the final design. Has anyone in the community had experience with Verizon and their SIP product? Were you able to get through the field trial successfully? What was the design that you used to get Asterisk certified with Verizon's network? Where I am at is that they want us to use an SBC. One engineer asked about Cisco Call Manager. I told them that basically if I can accomplish the same thing with a Linux box (routing box and sip proxy box) without having to spend money on SBCs or expensive Cisco gear, that is the route we would like to go. We are looking at the possibility of handling 140 concurrent calls... that is what they are designing on their end as well. So, I am asking the community for any input. I have read on here and seen on IRC that some in the community are successfully using Asterisk with Verizon SIP. Verizon was going to check and see if they have any notes about that and those particular setups. Can anyone help share any information or tidbits on how they were able to sucessfully work with Verizon? Thanks, -- Michael L. Young -- _ -- 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] SIP Debugging Information..
- Original Message - > From: "Howard Leadmon" > To: asterisk-users@lists.digium.com > Sent: Saturday, November 24, 2012 3:19:10 PM > Subject: [asterisk-users] SIP Debugging Information.. > > > I did a little googling, but didn't seem to find anything specific > to > answer the question. I am trying to debug some calls on an Asterisk > system > (AsteriskNow) that are dropping, and when the general logs didn't > nail > anything I turned on SIP Debugging on the trunk to the provider. > Basically the complaint is that when some call in, regardless of if > the call > is answered, or if Vmail answers it, it drops the calls in a matter > of > seconds. The strange thing is, that the system processes many > hundreds of > calls daily, but only a couple specific incoming callers are seeing > the > drops. I would have thought a NAT issue, but why does this only > affect a > specific group of incoming callers, the rest go about their business > just > fine. I think thinking bandwidth.com is mucking something up, but > again I > have no specific proof one way or another, so why the debugging. > > When one of the problem callers is dropped, in the SIP debugging I > see: > > chan_sip.c: Scheduling destruction of SIP dialog > '285991942_79966325@192.168.27.72' in 6400 ms (Method: BYE) > > > Is this the remote end (ie bandwidth.com) dropping the call, or is > the local > Asterisk server dropping the call? [snip] > --- > [Nov 23 15:43:13] VERBOSE[5127] chan_sip.c: > <--- SIP read from UDP:216.82.224.202:5060 ---> > BYE sip:4104159270@10.98.4.36:5060 SIP/2.0 > Record-Route: > Record-Route: > Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKe902.53bde7e.0 > Via: SIP/2.0/UDP 67.231.4.93;branch=z9hG4bKe902.32697e93.0 > Via: SIP/2.0/UDP 192.168.27.72:5060;branch=z9hG4bK0bBac8c2c3cb90659df > From: ;tag=gK0b66d829 > To: ;tag=as0850c6db > Call-ID: 285991942_79966325@192.168.27.72 > CSeq: 297 BYE [snip] If I am reading this right, it looks like a BYE is coming in from the far end, Bandwidth.com. Michael (elguero) -- _ -- 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] * Waiting for asterisk to shutdown .............
- Original Message - > From: "Joseph" > To: asterisk-users@lists.digium.com > Sent: Saturday, November 24, 2012 12:54:12 AM > Subject: [asterisk-users] * Waiting for asterisk to shutdown . > > I'm running asterisk on a small box, > Intel-R-_Atom-TM-_CPU_330_@_1.60GHz > and when I try to restart the asterisk it fails. > > /etc/init.d/asterisk restart > * Caching service dependencies ... [ ok ] > * Killing wrapper script ... [ ok ] > * Stopping asterisk PBX gracefully ... > > * Waiting for asterisk to shutdown > . > * Failed. > > When I run /etc/init.d/asterisk status > I get: "* status: started" > > At this point I have to kill the process ID > "zap" it (/etc/init.d/asterisk zap) > and restart it. > > Why asteriks can not shut down properly? > How can I monitor this process and restart it? What version of Asterisk are you running? Michael (elguero) -- _ -- 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] Intruder
- Original Message - > From: "Felix Vazquez" > To: asterisk-users@lists.digium.com > Sent: Friday, November 16, 2012 11:20:46 AM > Subject: [asterisk-users] Intruder > I am in the asterisk CLI and can see an unidentified caller trying > the make calls out of the asterisk system. How do I stop them? How > do I identify them and how can I see how the go in? > This is an example of what I would see: > NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call from '' to > extension '90111235551212' rejected because extension not found. I would recommend you read README-SERIOUSLY.bestpractices.txt, top level of source code. Another thing you can do is turn on security logging if you are using Asterisk 10/11. Take a look at logger.conf. It may provide you with some extra information on who is trying to make the call. Take a look at this page: https://wiki.asterisk.org/wiki/display/AST/Important+Security+Considerations I would recommend using fail2ban as well. Michael (elguero) -- _ -- 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] Error: astobj2.c: refcount -1 on object
- Original Message - > From: "Ishfaq Malik" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, November 14, 2012 9:25:37 AM > Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object > > Thanks for the advice but that's not really a quick and easy option > for > us. We would not be able to upgrade to another version without doing > full regression testing on the candidate upgrade version and we've > been > using this version for at least half a year and this is the first > time > we've had this crash and this error. > > Also, just upgrading doesn't enlighten me to what is going on to > cause > this error. Sorry, I thought I was offering a quick and easy option. Since, this is a minor release upgrade, there shouldn't (won't say there won't) be any changes to cause you problems. But, I understand you have to follow your procedures before upgrading. Software is not perfect. Since this is not happening all the time, it may take a while for you to try and figure out what is wrong; that is if you can reproduce it. Therefore, in my opinion, I think you would be better off using your time to consider upgrading especially with a lot of bugs and security updates being in the latest version. Use that time to run through your regression testing. Anyways, if you want to go the path to try and figure out what caused this, I beleive you will need to look at the following information: https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging Hope that helps, Michael (elguero) -- _ -- 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] Error: astobj2.c: refcount -1 on object
- Original Message - > From: "Ishfaq Malik" > To: asterisk-users@lists.digium.com > Sent: Wednesday, November 14, 2012 4:05:21 AM > Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object > > Hi > > I'm using 1.8.7.0. This morning I got an alert telling me > > Asterisk on exited on signal 11. Might want to take a > peek. > > When I had a look at the logs I can see a lot of errors like > > ERROR[14924] astobj2.c: refcount -1 on object 0x2aaad4069068 > ERROR[14924] astobj2.c: refcount -2 on object 0x2aaad4069068 > > All the way up to > > ERROR[14924] astobj2.c: refcount -2004 on object 0x2aaad4069068 > > Everything up to this point was completely normal. > > Does anyone know what this error means and what causes it? I would recommend updating to the latest version. We are up to 1.8.18 and 1.8.19 is around the corner. There have been a lot of bug fixes and you might find that whatever caused this issue is already fixed. Michael (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashing when trying to start a Jabber session with ejabberd
- Original Message - > From: "Patrick Lists" > To: asterisk-users@lists.digium.com > Sent: Tuesday, November 13, 2012 4:35:54 AM > Subject: Re: [asterisk-users] Asterisk crashing when trying to start a Jabber > session with ejabberd > > On 11/13/2012 12:11 AM, Phil Reynolds wrote: > [snip] > > It turns out to be a known issue: > > > > https://issues.asterisk.org/jira/browse/ASTERISK-19532 > > > > ... and can be fixed by applying the patch at: > > > > https://issues.asterisk.org/jira/secure/attachment/43441/xmpp_no_crash_with_ejabberd.patch > > > > I will file the details with Debian too... > > Is it an omission that this fix has not been applied to the 11 tree? > From the looks of ASTERISK-19532 it seems that the fix has only been > applied to 1.8 and 10. > If you click on the link for ASTERISK-19532, there is a tab in the Activity section labeled "Subversion". It shows that the patch was applied to 1.8, 10, 11 and trunk. Michael (elguero) -- _ -- 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] 11.0.1: more sip registry woes
- Original Message - > From: "sean darcy" > To: asterisk-users@lists.digium.com > Sent: Wednesday, November 7, 2012 9:20:58 AM > Subject: Re: [asterisk-users] 11.0.1: more sip registry woes > > On 11/06/2012 09:45 PM, Michael L. Young wrote: > > - Original Message - > >> From: "sean darcy" > >> To: asterisk-users@lists.digium.com > >> Sent: Tuesday, November 6, 2012 7:51:04 PM > >> Subject: [asterisk-users] 11.0.1: more sip registry woes > >> > >> Upgrade to 11. This worked on 10.X.X > >> > >> sip.conf: > >> > >> register=>:@nyc.teliax.net > >> > >> telnet nyc.teliax.net 5060 > >> Trying 8.14.120.23... > >> Connected to nyc.teliax.net. > >> Escape character is '^]'. > >> > >> sip show registry > >> Hostdnsmgr Username > >> Refresh > >> StateReg.Time > >> nyc.teliax.net:5060 N > >> 120 > >> Unregistered > >> 1 SIP registrations. > >> > >> > >> Nothing on the cli to show any problems. > >> > >> teliax says no problems on their end. > >> > >> In 10 if I wasn't registered I got lots on registration failed > >> messages. > >> > >> Added this to sip.conf: > >> > >> registertimeout=20 ; retry registration calls every 20 > >> seconds (default) > >> registerattempts=0 ; if 0 try forever > >> > >> which is supposed to be the default anyhow. > > > > I am registered without any problems to nyc.teliax.net. > > > > How is your peer definition set in sip.conf? Try turning verbosity > > up on the console and also "set sip debug" on for your peer in > > order to see the communication between your server and Teliax. > > Hopefully, that will provide some clues as to why you are not > > registering. > > > > Michael > > (elguero) > > > > Are you running 11.0.1? > > sean I am running trunk which is essentially 11.0.1 since there have not been any changes made in that area of the code since it was branched. Michael (elguero) -- _ -- 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] 11.0.1: more sip registry woes
- Original Message - > From: "sean darcy" > To: asterisk-users@lists.digium.com > Sent: Tuesday, November 6, 2012 7:51:04 PM > Subject: [asterisk-users] 11.0.1: more sip registry woes > > Upgrade to 11. This worked on 10.X.X > > sip.conf: > > register=>:@nyc.teliax.net > > telnet nyc.teliax.net 5060 > Trying 8.14.120.23... > Connected to nyc.teliax.net. > Escape character is '^]'. > > sip show registry > Hostdnsmgr Username Refresh > StateReg.Time > nyc.teliax.net:5060 N > 120 > Unregistered > 1 SIP registrations. > > > Nothing on the cli to show any problems. > > teliax says no problems on their end. > > In 10 if I wasn't registered I got lots on registration failed > messages. > > Added this to sip.conf: > > registertimeout=20 ; retry registration calls every 20 > seconds (default) > registerattempts=0 ; if 0 try forever > > which is supposed to be the default anyhow. I am registered without any problems to nyc.teliax.net. How is your peer definition set in sip.conf? Try turning verbosity up on the console and also "set sip debug" on for your peer in order to see the communication between your server and Teliax. Hopefully, that will provide some clues as to why you are not registering. Michael (elguero) -- _ -- 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] Case-sensitivity of Dialplan variables.
- Original Message - > From: "Matthew Jordan" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, October 3, 2012 12:17:56 PM > Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables. > > From Mark's original e-mail: > > "Some of you might be eager to propose a configuration option to > decide which it should be. I'm sick of having hundreds of options > in Asterisk to slightly tweak the behavior one way or another. This > needs to go one way or the other, not be configurable." > > I can't overstate how much I agree with this. A configuration option > to > 'tweak' the behavior in pbx.c is much more likely to introduce > problems than > solve them. If a clear consensus cannot be reached, I'd err on the > side > of doing nothing than put in yet another configuration option. I agree that a configuration option is not the solution. I am not seeing what the big deal is. Software changes between major releases. Someone is not going to, or at least they shouldn't if their livelihood depends on it, upgrade their machines without doing the proper preparation for upgrading. That means reading the UPGRADE.txt file and outlining what needs to be done to upgrade their system if there are features they need in the new version or simply want to be on the latest version. Then they should test those changes as well before putting it into production. We are probably a year away from seeing a release for the version of Asterisk where this change would occur. We are two years away from an LTS version of Asterisk. So, I think there would be plenty of time for evaluation and testing to be performed by those affected. Especially, as in the case of what Raj mentioned at the beginning of his prior email, not too many people may even be affected by this change just like he won't be. Michael L. Young (elguero) PS: If you can't tell, I am really for this change and doing so without any configuration options :) -- _ -- 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] Case-sensitivity of Dialplan variables.
- Original Message -
> From: "Ira"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Wednesday, October 3, 2012 3:21:50 AM
> Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
>
> At 07:59 PM 10/2/2012, you wrote:
> >
> >While true that most users are probably not programmers, most people
> >administering Asterisk would be system / network admins,
> >correct? System admins and networking admins are used to working in
> >environments such as Linux where variables and file names are case
> >sensitive.
> >
> >If someone is moving from a GUI interface to CLI, then they
> >would/should know that case sensitivity is important and therefore
> >the change shouldn't pose a problem.
>
> I'm not a system / network admin, at least not for Linux. I have one
> Linux machine, it runs Asterisk and Samba. I can usually make
> Asterisk do what I want. Samba works but I have little to no idea
> why. I run "yum update" occasionally and I run V11 trunk or whatever
> the proper name would be for the development version.
I can think of some situations where case sensitivity could be a problem. I
hope I am not out in left field with my thinking. Asterisk can be found in
companies that have several offices. Asterisk could be used in a cluster.
Asterisk may be administered by many different folks at a company and probably
more than one Asterisk box. If those individuals are expecting the variables
to be case sensitive, it becomes a problem trying to debug problems in the dial
plan. They may not know that an individual in one office is doing things one
way because they are not expecting variables to be case sensitive while another
individual is expecting things to be case sensitive. It really can create a
lot of trouble and confusion in bigger deployments versus a single individual
administering his own box.
> If there was a compiler and declared variables then case makes
> perfect sense. Without that, I'd never get a C program to work.
>
> I know people want case sensitivity, it's the "right" way to do it,
> but how does it help Asterisk?
This helps Asterisk by following a more or less established standard that
everyone expects. I believe that this case-insensitivity in the dial plan
actually came as a surprise to some who had never stumbled across it before.
Again, those with experience in unix/linux environments have been trained that
variables are case sensitive and they do not have to be programming in C to
know that.
> Does anyone have configurations that would be broken by case
> insensitivity?
Some people might have broken dial plans and that is why this was brought up on
the list in order to gain attention and feedback. But, it will only break for
the next release. It won't affect current releases. Instead, Mark is planning
on documenting the current behavior on the Asterisk wiki. From what I am
observing so far, it looks like it may only affect a small number of people.
My feeling is that the majority may have already been using variables as if
they were case sensitive already. That was how variables were documented on
the Asterisk wiki... as being case sensitive.
> If not, then what is the upside of enforcing case sensitivity?
The upside is that we have consistency. This helps to keep bug reporrts to a
minimum and in my opinion helps the end user not to create problems for
themselves. The example mentioned in the issue being worked on, is say, an
application is expecting the variable ${MIXMONITOR_FILENAME}. A user thinks,
"Hey, the dial plan is case insensitive" and uses ${mixmonitor_filename} or
${MixMonitor_FileName} to set the file name. They find out that the variable
is being ignored. They later check the variable ${MIXMONITOR_FILENAME} (notice
all uppercase) in the dial plan and it shows him that it is set. They then
think there is a bug in Asterisk... well, the problem is that they didn't set
the variable according to what app_mixmonitor is expecting. The application IS
case sensitive when it comes to variables. So, this is the confusion that can
be caused by having one part of Asterisk be case sensitive and another part of
Asterisk NOT be case sensitive.
I hope this explanation helps those reading this to understand better what is
trying to be resolved here. At least, this is the way I am understanding the
reason for the proposal presented to the list.
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Re: [asterisk-users] Case-sensitivity of Dialplan variables.
- Original Message - > From: "Ira" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, October 2, 2012 8:11:32 PM > Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables. > > > Given that many of the users were not programmers and didn't likely > grow up in a case sensitive world I'd also vote for case > insensitivity. I fall into that category, I grew up with dBase, > Clipper and VB and case issues get me all the time when I program in > C. > > Allowing case insensitivity does not stop someone from using case > consistently and While I guess there could be a reason why you'd want > to use the word hash in the forms hash, Hash and HASH and have them > be 3 different items, I'm guessing that the people trying to get > their feet wet moving from Asterisk-Now to Asterisk would be confused > to say the least if someone did that in example code. While true that most users are probably not programmers, most people administering Asterisk would be system / network admins, correct? System admins and networking admins are used to working in environments such as Linux where variables and file names are case sensitive. If someone is moving from a GUI interface to CLI, then they would/should know that case sensitivity is important and therefore the change shouldn't pose a problem. Just some thoughts in regards to the concerns brought up. Michael L. Young (elguero) -- _ -- 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] Case-sensitivity of Dialplan variables.
- Original Message -
> From: "Vladimir Mikhelson"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, October 2, 2012 9:02:18 PM
> Subject: Re: [asterisk-users] Case-sensitivity of Dialplan variables.
>
>
> On 10/1/2012 4:15 PM, Mark Michelson wrote:
> >
> > What I plan to do, no matter which way the vote goes, is to
> > document
> > on the wiki how things currently behave in Asterisk, to include the
> > example I gave above (or something similar anyway). Depending how
> > the vote goes, I will make the necessary code changes in Asterisk
> > trunk. I will document the behavior change both in UPGRADE.txt and
> > on the wiki.
> >
> > When considering which way you lean, consider that we really don't
> > have much of a precedent to go on. For instance, dialplan
> > applications
> > are case-insensitive ("answer" and "Answer" and "ANSWER" are all
> > the
> > same). Dialplan functions, on the other hand, are case sensitive
> > ("HASH" would be evaluated properly but "hash" would not). My
> > personal
> > opinion is that all variable evaluations should be case-sensitive.
> > I don't feel all that strongly about it though and could easily be
> > swayed the other way if people respond overwhelmingly in
> > opposition.
> >
> First you need to consider compatibility with currently supported
> packages which include auto-generated dial plans like AsteriskNow,
> PIAF,
> etc. If you plan to break their functionality you need to at least
> coordinate your move with the maintainers.
This change would go into trunk, the development line of code. So, I am not
sure that a coordinated effort would need to be made here. As with every major
release, maintainers of external packages usually check the UPGRADE.txt. There
almost always will be something to change between major versions in order to
keep in step with the new release.
> Then you may want to consider backwards compatibility with packages
> still widely used but not actively supported any more like Trixbox.
> Maybe not the best example as their WEB site says, "This is the
> current
> stable release based on Asterisk 1.6."
Again, I am not sure there is a need for backwards compatibility. This change
would go into Asterisk 12, which is not LTS. I would think that most packages
would be focused more on working with the LTS version of Asterisk. I could be
wrong though. Given that Asterisk 11 is the current LTS that will be probably
be released soonish out of beta, the next LTS version is a couple of years away
giving plenty of time for people to make the necessary changes.
> If you really want to make it not settable (and this is big, not a
> minor
> change, if I were you I would definitely make it settable) then I
> would
> go with case-insensitive as it allows for various custom notations,
> e.g.
> Hungarian notation in naming of custom variables without a later
> painstaking investigation whether "nCallID" is equal to "nCallId" or
> not. Consider the fact, most of the dial plan debugging happens in
> the
> logs or in the Console Screen. Someone may want to spell "nCALLID"
> just
> to be able to see the difference between Latin "l" and "I" where the
> first one is "L" lower case and the second one is "i" upper case.
I didn't quite follow this logic. Your example, in my mind, would actually be
easier to debug with this change.
If you know that variables are case sensitive, you know that you have to check
for a typo in your variable name if you are not getting what you were
expecting. Here, in my email client, the "l" and "i" are very distinct as well
as the console I work in.
Just my thoughts on the above concerns presented.
Michael L. Young
(elguero)
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Re: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)
- Original Message - > From: "Thorsten Göllner" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Monday, June 18, 2012 11:52:15 AM > Subject: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging > (mysql, odbc) > > > /etc/odbcinst.ini > > [MySQL] > Description = MySQL ODBC MyODBC Driver > Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so > FileUsage = 1 Try adding: "Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so" Adjust the path according to where this file can be found on your system. > So here are the config file for asterisk. > > /etc/asterisk/res_odbc.conf > - > [mysql] > enabled => yes > dsn => MySQL-asterisk > username => asterisk > password => qpalym > pre-connect => yes For MySQL, I think you also need: "backslash_is_escape => yes" Give those two things a try and see if that helps. Regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
- Original Message - > From: "Jayesh Labade" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, May 25, 2012 2:09:58 AM > Subject: Re: [asterisk-users] Asterisk MixMonitor starts recording 44 > bytes file > Hello Michael, > Thanks a lot for your immediate help. After applying patch MixMonitor > started works normally, > I can understand that this can be Happen in asterisk 10.4 but as a > stable and Long support version 1.8.12.0 this should not happen. I > got same error in both version. > Anyways this patch solved my problem. Jayesh, Glad to hear that the information helped figure out what was going on and also provided a fix. In the 1.8 line, this has been fixed as well and will be in future releases. In an ideal world, there would be no bugs in software. LTS doesn't mean "bug free". It means that it will be supported over a longer period of time which should result in more real world use and more bug fixing resulting in a more stable product with time. Regards, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk MixMonitor starts recording 44 bytes file
- Original Message - > From: "Jayesh Labade" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Thursday, May 24, 2012 4:10:29 PM > Subject: [asterisk-users] Asterisk MixMonitor starts recording 44 > bytes file > Hello All, > I have installaed asterisk 10.4 in my machine. Now suddenly > MixMonitor application starts generating 44 Bytes of Recording file. > Is this new tye of Bug? Help me.. > Best Regards, > Jayesh Labade Jayesh, Is this machine x86? There was a bug that was recently fixed and should show up in 10.5. https://issues.asterisk.org/jira/browse/ASTERISK-19727 Regards, Michael -- _ -- 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] All circuits are busy now on outgoing trunk call
> [0K > <--- SIP read from UDP:192.168.9.251:5060 ---> > SIP/2.0 404 Not Found > > Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 > > To: > ;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 > > From: "pbxserver" ;tag=as66c75bd7 > > CSeq: 102 INVITE > > Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 > > Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) > > Content-Length: 0 I think the "404 Not Found" being returned from the server is a clue as to what the problem is. Michael L. Young (elguero) -- _ -- 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] No valid transports available, falling back to 'udp'.
- Original Message - > From: "David C Klaverstyn" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Sunday, February 12, 2012 7:02:15 PM > Subject: [asterisk-users] No valid transports available, falling back > to 'udp'. > Hi All, > I just installed Asterisk 10.1.1 and on sip reload I see the > following two errors. > WARNING[3665]: chan_sip.c:29242 reload_config: No valid transports > available, falling back to 'udp'. > == Using SIP CoS mark 4 > WARNING[3665]: chan_sip.c:27839 build_peer: 'tcp' is not a valid > transport type when tcpenabled=no. If no other is specified, the > defaults from general will be used. > I don’t understand this as I have in sip.conf > udpbindaddr=0.0.0.0 > I’ve also tried > udpbindaddr=192.168.13.7 > and > udpbindaddr=192.168.13.7:5060 > with the same results. > Is there another setting that I am missing. I have not made any > changes to the default file settings, well none that I remember. This WARNING message is letting you know that TCP is not available as a transport. So, it is falling back to UDP as the transport. Unless you are trying to use TCP, it is a harmless message. The message should probably be changed or only displayed when a particular setting is enabled since, as it is, it is a bit misleading. If you don't care to use TCP and have it disabled in the configuration file, you shouldn't be alerted to the fact that Asterisk is unable to use TCP. Feel free to open up an issue as Paul directed in his prior message. Regards, Michael Young (elguero) -- _ -- 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] TDM400 FXO stopped working
- Original Message - > From: "Remco Barendse" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, September 23, 2011 5:27:27 AM > Subject: [asterisk-users] TDM400 FXO stopped working > Hi list > > I have 2 servers with a TDM400 card, port 1 populated by an FXO (red) > module), port 4 populated with an FXS module. I am using dahdi > linux and tools 2.5.0.1. The servers are running CentOS 4 and the > other > box CentOS 6. > > Both modules have been working fine but recently stopped working, when > i > start dahdi with just FXS enabled everything is fine. > > This is the error i get : > Loading DAHDI hardware modules: > wctdm: [ OK ] > > Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 1: Invalid > argument > (22) > Selected signaling not supported > Possible causes: > FXO signaling is being used on a FXO interface (use a FXS > signaling variant) > RBS signaling is being used on a E1 CCS span > Signaling is being assigned to channel 16 of an E1 CAS span > [FAILED] > > > This is in my system.conf : > fxoks=1 > echocanceller=mg2,1 > # channel 2, WCTDM/4/1, no module. > # channel 3, WCTDM/4/2, no module. > fxsks=4 > echocanceller=mg2,4 > > # Global data > > loadzone = nl > defaultzone = nl I think the clue is actually right there in the error message. You say that port 1 is an FXO module? Then your signaling is set wrong. The signaling should be fxsks. For port 4, it should be fxoks. Remember, that in the configuration files, the signaling option used is opposite of what the module is. Regards, Michael Young (elguero) -- _ -- 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] Polycoms rebooting themselves
- Original Message - > From: "Mike Diehl" > To: asterisk-users@lists.digium.com > Sent: Tuesday, August 30, 2011 5:13:22 PM > Subject: Re: [asterisk-users] Polycoms rebooting themselves > > Well, we've taken the time to check out the wiring. It's only 3 > years old and > looks like the people who did it knew what they were doing. Nice > work. > > Rebooting the cable modem, router, and switch didn't fix the problem. > > Also, we had an instance today where ALL of the phones went down > within > minutes of each other. The Internet connection was still active. > > Looks like more often than not, all of the phones die at the same > time. > > Any other ideas? > > Mike. How "latest(ish)" is the firmware? I see in the release notes for 3.3.2 under corrections: 61147: SoundPoint IP 331, 335, 450, 550, 560, 650, 670: SoundStation IP 5000: Phone reboots when a GET request is sent to the phone to /TA/getParam?paramName=reg.1.ringType. 68063: Phone reboots when DHCP failover occurs. (I know you have 335s but someone else mentioned they had issues with 550s) 70988: SoundPoint IP 550: Phone when powered by external AC power reboots during playing of certain audio on full volume. Just some things that came to mind when I saw your email. I had just recently reviewed the release notes and they were fresh in my mind. Hope you find a fix soon. Regards, Michael (elguero) -- _ -- 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] Issue 0019268 Patch Asterisk
- Original Message - > From: "Patrick Lists" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, July 8, 2011 1:58:36 PM > Subject: Re: [asterisk-users] Issue 0019268 Patch Asterisk > > On 07/08/2011 07:32 PM, Mark Rosedale wrote: > > > * channels/sig_pri.c: PRI early media won't ring. And another way > > to pass early media. Don't indicate that there is inband > > information present, just assume that the B channel is > > connected. > > * Restore clearing the dialing flag Rx squelch unconditionally > > when a PROCEEDING message comes in. (closes issue #19268) > > Reported by: tbsky Patches: issue19268_v1.8.patch uploaded by > > rmudgett (license 664) Tested by: tbsky > > I looked at the 1.8.5-rc1 code. I don't see that patch in the > 1.8.5-rc1 > branch. Did I miss something? > > >> http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/channels/sig_pri.c?view=markup > > Regards, > Patrick > Patrick, The patch was merged in to the 1.8 branch on 5/13/2011 as revision 318783 (http://svnview.digium.com/svn/asterisk?revision=318783&view=revision). 1.8.5-rc1 was tagged on 06/29/2011 (http://svnview.digium.com/svn/asterisk/tags/1.8.5-rc1/?view=log). So, it would be in there since the tag is copied from the 1.8 branch at revision 325707. Looking at the code... it is definitely in there. Line 5503 was moved to line 5505. Michael (elguero) -- _ -- 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] ConfBridge - Failed to find a bridge technology to satisfy capabilities
- Original Message -
> From: "Chris Maciejewski"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Thursday, May 19, 2011 9:39:57 AM
> Subject: [asterisk-users] ConfBridge - Failed to find a bridge technology to
> satisfy capabilities
>
> Hi,
>
> I am trying to use ConfBridge application, but it throws "Failed to
> find a bridge technology to satisfy capabilities 0x4 (ulaw)" error.
> Please see console output below.
>
> -- Executing [501@services:9] ConfBridge("SIP/OpenSER-0005",
> "1001") in new stack
> [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404
> join_conference_bridge: Trying to find conference bridge '1001'
> [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed
> to find a bridge technology to satisfy capabilities 0x4 (ulaw)
> [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368
> destroy_conference_bridge: Destroying conference bridge '1001'
> [May 19 13:36:05] ERROR[7452]: app_confbridge.c:435
> join_conference_bridge: Conference bridge '1001' could not be
> created.
>
I wonder if this recent commit to the 1.8 branch would help fix this issue at
least with 1.8.
Author: twilson
Date: Thu May 19 18:28:13 2011
New Revision: 319920
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=319920
Log:
Revert part of a change to the bridging API code
The capabilities used in the bridging API are very different than the
ones used for formats. When the conversion was made expanding the bit
width of codecs, the bridging code was accidentally accosted in ways
that it didn't deserve.
Modified:
branches/1.8/include/asterisk/bridging.h
branches/1.8/include/asterisk/bridging_technology.h
branches/1.8/main/bridging.c
As far as why svn trunk is stating that it cannot create the bridge, the debug
message is not very informative as to why it couldn't create the conference
bridge from what I could see briefly looking at the debug logs you posted in
another message.
Michael
(elguero)
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Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
- Original Message - > From: "Olle E. Johansson" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Wednesday, April 27, 2011 3:34:03 PM > Subject: [asterisk-users] Discussion: Are we ready to leave 1.4 behind? > > Friends, > > We have a discussion on asterisk-dev about the maintenance of the 1.4 > branch. According to the release plans, support for 1.4 was > scheduled to close in April 2011 - basically now. After that, only > security patches would be committed. This is already a delay from > the original plan published by Russell Bryant. > > Unfortunately, I think this is way too early. My feeling and > experience is that 1.8 is not ready for production in the > environments I work in - large scale installations. Customers are > not planning migration and all new installs are still 1.4. Tests > we've been doing with 1.8 has failed within just a short time and so > badly that customers has not paid me to spend any further time with > 1.8. > Whats the game plan to get 1.8 "ready for production"? To me, for which I say this with all respect, some are focusing still on 1.4 instead of getting 1.8 to the level that some of the members of the community are wanting to see. 1.4 has been very stable for a while. To the point that I only pay attention to security releases to be honest. It has been this way for quite a while now. I personally have been focused more on using 1.8 when I can, mainly on non-critical servers, yet I will admit that I have enough confidence in it now to use on main servers. Why? Because I want to get my production servers off of 1.4 and 1.6.2 due to new features. But, even if I didn't need or want the new features, the current state of 1.4 is excellent. If I don't ever make the move beyond 1.4, how can I contribute to a better product? By experimenting and not giving up at the first sign of trouble with the latest version, I feel that I can help to make 1.8 better which ultimately benefits me and the community. I would like to hear a game plan before we just say, yes, lets keep focusing on 1.4 and then we will decide a deadline to stop support. I am afraid that software is programmed by imperfect humans and there will always be a bug or two that crops up from time to time. Do we want to keep waiting until we feel it is "perfect"? One thing I have noticed, is that the bug fixes and patches being contributed for 1.8 and trunk are not being taken care of as quickly as it used to back in the early 1.4 days. My feelings are that it is because there have been too many releases to work on. Going back to focusing on just 1.8 and trunk, would go a long way to speeding up bug fixes to 1.8. Again, just my opinion. > Last time we went through this process with a LTS release (which we > did not know then) it took over one year before we had a stable > product to migrate away from 1.2 and jump on the 1.4 track. > Hopefully, with the help of community, we can move up to 1.8 late > this year or early next year. For me 1.8 is the focus, it's the LTS > release. > > Not having a supported 1.4 version from the Digium-hosted > repositories will mean that we will have to move to separate > repositories or branch off from the main track. I already maintain a > ton of subversion branches with various patches to 1.4 It takes a > lot of time to manage this version that is a fork from the main 1.4 > branch. I will soon have to start working with subversion branches > for 1.8 to create a compatible version for my customers to test, > since most of the patches is not part of 1.8. After a few years of > doing this, I know the work involved with managing code myself. > > The Digium team wants to go ahead and not support 1.4 any more, I > want to keep 1.4 open for normal bug fixes. What do you think? Was this really Digium's decision? You keep mentioning Digium and implying them as the evil one in all of this (perhaps I am just misunderstanding your tone in your emails and if I am, I sincerely apologize for this) when I seem to recall plenty of discussion around these time lines and it was the community who set the deadlines, not Digium. Digium is just trying to abide by the time lines outlined for them by the community. They have already been nice enough to extend the deadline in order to finish working on outstanding bug reports and patches. They have bills to pay too and have really tried to extend an olive branch to everyone in the community. There has been a lot of activity on the 1.4 branch lately. If I am wrong, I will gladly retract my comments. > > Kevin proposed that the community maintains the 1.4 branch without > support from the Digium team. I don't think that's a good solution, > but it may be the only solution. I haven't got the resources to > manage the 1.4 code myself, so I won't step forward as a maintainer > if I can't get proper funding. Anyone else out there that has the > time and resources to manage the code
Re: [asterisk-users] PRI D-channel bouncing
-- Michael L. Young Administrative Claim Service, Inc. | IT Manager 600 Main Street, Suite 5, Winchester, MA 01890 www.acsacc.com Phone 781-721-1998 - Original Message - > From: "Andrew Stewart" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Tuesday, August 10, 2010 9:33:45 AM > Subject: [asterisk-users] PRI D-channel bouncing > I need some help getting a system running for one of my company's > plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and > FreePBX 2.8.0.2. > > My D-Channel keeps bouncing. The telecom tech told me he thought that > I might be using the wrong sync source, and I think I might have been, > but I changed DAHDI system.conf to "span=1,1,0,ESF,B8ZS" (from > "span=1,0,0,ESF,B8ZS") and I am still having the same problem. > (Although, the FreePBX DAHDI page only allows me to select "0" in the > "Sync/Clock Source" field. "0" is the only option in the drop down.) > * > [r...@gch-asterisknow01 ~]# cat /etc/asterisk/chan_dahdi_groups.conf > ;; > ; Do NOT edit this file as it is auto-generated by FreePBX. All > modifications to ; > ; this file must be done via the web gui. There are alternative files > to make ; > ; custom modifications, details at: > http://freepbx.org/configuration_files ; > ;; > ; > > > ; [span_1] > signalling=pri_net > switchtype=national > pridialplan=national > prilocaldialplan=national > group=0 > context=from-pstn > channel => 1-15 Is the PRI coming from the telephone carrier? If so, shouldn't the signalling be pri_cpe? Michael L. Young (elguero) -- _ -- 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] Dropped Calls
> -Original Message- > From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of JR Richardson > Sent: Tuesday, March 30, 2010 6:55 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Dropped Calls > > > I've written about this issue several times, but have not yet found any > > solution to it. I am using asterisk 1.4.21.2 and zaptel 1.4.12. Phones > > are primarily Snom 300's but I also have a couple of headset phones > > connected to Grandstream HT286 SIP adapters. I have 8 offices, each has > > it's own asterisk server all running the same versions of asterisk and > > Zaptel. Only difference is that one office uses a Digium TDM 8-port > > card and the other branches use 4-port Rhino cards with only 2 ports in > > use. What happens is that periodically we will be in a call and the > > call will just drop. It's usually within the first couple of minutes of > > the call. The calls can be either incoming or outgoing. The phenomenon > > affects both the Snoms and the Grandstreams. Along with the dropped > > call issue, we periodically have a problem where a person we call or a > > person that calls in cannot hear the person in the our office, but the > > person in our office can hear the remote person fine. > > > > All of the phones are on the same physical network as the asterisk > > server. There is no NAT, no Firewall, VLAN, etc. between the phones and > > the server. I have tried running sip debugs on the calls, but on the > > off chance that my logs catch either a drop or a one-way audio, the sip > > debug looks like just a normal call. > > > > Is there any setting that might cause both one-way audio and dropped > calls? > > > > Thanks, > > Brent Davidson > > Join the club. I've experienced the same with various strains on > 1.4.x above 1.4.21.1 (not an issue with this one that I have seen). > This issue is truly random and debugging reveals nothing. I run an > all SIP environment with same results. My solution was to downgrade > to another version or switch to 1.2 or 1.6 depending on what features > I need for the system. > > Sorry I couldn't be of any help, but I feel your frustration. > > JR > -- > JR Richardson > Engineering for the Masses > Is there a chance that you are using Realtime at all? I am just curious because I was having problems with dropped calls as well and just discovered that it appears to be related to the database server. If for some reason on the database server there is a table lock (which I am investigating why) asterisk drops any PRI calls and SIP calls. Everything looked normal and the error messages never once suggest a problem with the database server or Realtime. I was looking everywhere else but at the Realtime until I stumbled across it. While doing some backups with FLUSH READ LOCKS to a slave machine, which I changed asterisk to use a few months back, I had dropped calls occur. I later confirmed that asterisk seems to hang / freeze during that period but once the database server releases the locks, asterisk continues to function without any problems. This started to occur when we had an increase in call volume and an increase in load on the db server. I was using Realtime for extensions, sip peers and CDR. I had turned off using realtime for CDR (which we don't really use anyway) and started to use a slave server instead of the master when performing some maintenance on the master db server. I left it that way since I was just using it for extensions and sip peers and that had cleared it up over the last few months until I ran my backup. Not sure that helps but it is worth a shot in mentioning to you. Regards, Michael Young (elguero) -- _ -- 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] Got SIP response 420 "Bad Extension" back from inphonex.com
Hello: New to asterisk and hoping to use for http://summitcamp.org research station. While trying to use with Inphonex I find that incoming calls drop after about one minute-- -- Got SIP response 420 "Bad Extension" back from 208.239.76.169 == Spawn extension (incoming-inphonex, 210, 1) exited non-zero on 'SIP/inphonex-095bf208' Found that I can use `*CLI> sip set debug peer inphonex` to see more information, such as-- <--- SIP read from UDP://208.239.76.169:5060 ---> SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 64.165.113.66:5060;received=64.165.113.66;branch=z9hG4bK121e8b66;rport=5060 From: ;tag=as111b0d1e To: Unknown ;tag=SDbapb901-2318a5d8 Call-ID: SDbapb901-ff4e360f8a8714144f03eb06aad237b5-gurpkk2 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 The best I can figure is that inphonex does not support session-timers because when I insert the following-- sip.conf |session-timers=refuse The calls do not drop. Question is simply whether this will haunt me elsewhere. Thanks, Andrew Using CentOS release 5.4 (Final) / asterisk16-1.6.0.17-1_centos5 Registered to Inphonex-- register => virtuser:passwd:virtu...@sip.inphonex.com:5060/DID [inphonex] username=xxx type=peer secret=xxx ; password used to login their website (same as in register =>) host=sip.inphonex.com fromuser=xxx fromdomain=sip.inphonex.com context=incoming-inphonex ; context to be used in extensions.conf for inbound calls from inphonex canreinvite=no insecure=invite ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
Quoting Tim Nelson : > Do you have any sort of site/mailing list/etc setup to facilitate > this group? I'd be interested in attending such a meetup in the > future. http://www.tcaug.net/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Not Dialing 9
Quoting "Thczv F. Thczv" : > When I set up my Asterisk box at home I didn't want to have to dial 9 > to dial off premises, so I gave all my local phones three digit > extensions with this format: 1[1,0]*. My thought is that there are no > area codes that start with 0 or 1, so if I use those numbers, I can > create 20 local extensions that can be dialed with 3 digits, and not > have to use a timeout when dialing long distance. If I dial 1, then > anything other than 0 or 1, Asterisk knows I am dialing long distance. > If I start with any number other than 1, Asterisk knows I am dialing > a local or local toll call. In North America: 0 is the intra-lata operator 00 is the inter-lata operator 0+ will be an operator assisted call 11xx is used for the rotary dial equivilant of *xx on many central office switches. Assuming you are not using rotary dial, I generally use 4 digit extensions with the 11xx format for the same reason you suggest. --Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicepulse down
Quoting Fred Posner : > Starting around 10:00 AM EST. > > All services from them whether I connect by IP or DNS (both east coast > and west). Anyone else? Yes, I'm experiancing the same problem. Their www.voicepulse.com and connect.voicepulse.com seem to be offline as well. --Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Seemingly easy question: NPA/NXX
International numbers are variable length, so the timeout applies for those. North American "National" numbers are a fixed length. Generally, the phone company will collect 7, 10 or 11 digits for North American numbers. For example, I live in Minneapolis, MN. My number is 612-xxx-. I have free calling to 612, 651, 952, 763 and a few numbers in 507 and 320. If I dial 1, the phone company will collect 10 more digits. (The call may or may not go through if I dial 1+ a 10-digit local number depending on the carrier. MN regulations prohibit charging for local calls dialed as toll) If I dial 612, 651, 952, 763, 507 or 320, the phone company will wait for the remaining 7 digts as there are no numbers within area code 612 that start with those digits. Anything else will only be collected as 7 digts and assumed to be 612. Because of that, I can't dial a california number (for example), without dialing it as 1+. I wouldn't call it "fancy", the phone company just knows what is a valid local number for you. Making a digit map in an ATA isn't that hard, you just need to think about what you want it to do. If you want to permit 10 digit dialing without the 1+ for long distance *and* support 7 digit local dialing, you'll need a timeout. There are also the N11 numbers, which of course should stop collecting after the second "1". --Shane Quoting Karl Fife <[EMAIL PROTECTED]>: > Question: > How does the local Telco know you're done dialing a seven digit number? > Easy you may say: If your dial string begins with 1, the parser expects > 11 digits total, otherwise seven, 011 is international. > > The reason suspect it's more complex is that: > 1) International numbers can vary widely in length and > 2) Our local analog Telco will route a ten digit NANP numbers with no > leading 1 and with no terminator--seemingly instantly > > Obviously this could be done with 'timeouts'--implicitly 'sending' > after a delay. But it works so well I suspect there's more logic in > there. For example I have dozens of ATA's provisioned with timeouts, > and I find it difficult or impossible to replicate the Telco dialing > experience (Either the delay is too long, or you have frequent 'reorder' > tones because it 'sent' before you were finished). > > Therefore I assume that there is something more 'fancy' going on. Can > someone validate, debunk or clarify this? > > Theory 1 > Is it all done with timeouts, but they're CONDITIONAL timeouts. > i.e. give a LONG timeout if the number: > -did not start with a 1 and is still shorter than 7 digits, > -started with a 1 and is still shorter than 11 digits > -started with a 011 and is shorter than the theoretical international > minimum lenght > > Theory 2 > As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. > Therefore the switch could easily determine(without the leading 1) if > your first three digits were an NPA or just an NXX (exchange). They > were nationally unambiguous. Now that's no longer true. STILL, it > could be possibleto consider all known valid NPA's and exchanges so they > can determine via context what you're trying to do, and thereby optimize > the dialing experience? > > Can anyone speak to this? I would very much appreciate any knowledgable > input. > > -Karl > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadsoft Sip provider
Quoting Gustavo A Gonzalez <[EMAIL PROTECTED]>: > I am looking for a sample sip configuration of a SIP provider that runs > Broadsoft VoIP switch. This is what I use: register => 3115552368:abcdefghijklmnop:[EMAIL PROTECTED]/3115552368 [broadworks] type=peer host=1.2.3.5 dtmfmode=rfc2833 outboundproxy=1.2.3.4 fromdomain=1.2.3.5 fromuser=3115552368 username=3115552368 authname=3115552368 secret=abcdefghijklmnop canreinvite=no disallow=all allow=gsm allow=g726 allow=ulaw qualify=yes insecure=port,invite context=inbound ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adit 600 password reset
Quoting Doug Lytle <[EMAIL PROTECTED]>: > C F wrote: >>> >> >> Then there is basicly no way to do this besides for cracking it? I >> > > Not that I am aware of, no. This subject went around several years > back. They also talk about brute forcing the password as well. As far > as I recall, nobody came back saying they were successful. > >> have already figured out the username, now I just need to figure out >> the password. What is a good screen automation program that can >> bruteforce this for Windows? I had the same problem with one of mine. I smply forgot the password. I seem to recall that the adit had a flaw in it, where it was obvious by the error message returned if you had the correct length username and password, which should make your brute-force attempt much easier. --Shane ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium T1 Card Crashing Server (Dell 2950)
> BUG: soft lockup detected on CPU#1! > [] softlockup_tick+0x96/0xa4 > [] update_process_times+0x39/0x5c > [] smp_apic_timer_interrupt+0x5b/0x6c > [] apic_timer_interrupt+0x1f/0x24 > . You don't happen to be running a XEN Kernel are you? I saw this problem while running CentOS 5.1 XEN kernel and if you search their bug tracking system you will see some reports about this bug. A search on google revealed some possible solutions. This was the first thought that came to my mind when I saw this. Regards, Michael L. Young (elguero) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Dropping Calls
# channel 1, WCT1, unhandled for now # channel 2, WCT1, unhandled for now # channel 3, WCT1, unhandled for now # channel 4, WCT1, unhandled for now # channel 5, WCT1, unhandled for now # channel 6, WCT1, unhandled for now # channel 7, WCT1, unhandled for now # channel 8, WCT1, unhandled for now # channel 9, WCT1, unhandled for now # channel 10, WCT1, unhandled for now # channel 11, WCT1, unhandled for now # channel 12, WCT1, unhandled for now # channel 13, WCT1, unhandled for now # channel 14, WCT1, unhandled for now # channel 15, WCT1, unhandled for now # channel 16, WCT1, unhandled for now # channel 17, WCT1, unhandled for now # channel 18, WCT1, unhandled for now # channel 19, WCT1, unhandled for now # channel 20, WCT1, unhandled for now # channel 21, WCT1, unhandled for now # channel 22, WCT1, unhandled for now # channel 23, WCT1, unhandled for now # channel 24, WCT1, unhandled for now # channel 25, WCT1, unhandled for now # channel 26, WCT1, unhandled for now # channel 27, WCT1, unhandled for now # channel 28, WCT1, unhandled for now # channel 29, WCT1, unhandled for now # channel 30, WCT1, unhandled for now # channel 31, WCT1, unhandled for now # Global data loadzone = uk defaultzone = uk span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 My zapata.conf file looks like this: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 musiconhold=default rxgain=0.0 txgain=0.0 immediate=no overlapdial=yes callgroup=1 pickupgroup=1 pridialplan=unknown faxdetect=incoming prilocaldialplan=unknown group=0 context=from-zaptel callerid=asreceived switchtype = euroisdn signalling = pri_cpe channel => 1-15,17-31 ;Include genzaptelconf configs ;#include zapata-auto.conf ;Include AMP configs ;#include zapata_additional.conf Kind Regards, Richard Young Intrintech Limited [EMAIL PROTECTED] 111 Cannon Street London EC4N 5AR Phone: 0845 644 2918 All orders placed or confirmed via email are automatically bound by our Terms and Conditions Registered in England and Wales number 4488657. VAT Registered 799 1247 79 Registered Address: Global House, 2 Crofton Close, Lincoln, LN3 4NT ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Show Callee name on Display
It's all priced by quantity of each feature you license, number of users, number of concurrent calls, things like that. Previously it only ran on Solaris. It now also runs on Linux. I wasn't involved with our initial purchase, but I couldn't imagine you could have a working system for less that 100k (not including hardware). Normal broadworks systems include: 2 Application Servers N+1 Network/Routing servers N+1 Media servers In addtion to the software, you'd need to purchase the hardware and OS to run it on. It will do SIP or MGCP on the user side and SIP on the back-end/PSTN side. It doesn't support any telephony hardware directly (nothing like zaptel). It just does SIP or MGCP. You'd need to connect to something that will get you back to the PSTN (either your own hardware or a provider) Quoting Seysan <[EMAIL PROTECTED]>: > Hello, > > Does anyone Knows the price of the Broadworks? > > any idea? > > Seysan > --Shane ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Show Callee name on Display
I don't think that Asterisk currently sends a remote-party-id to the called party. That would proably have to be added to the sip channel. It *does* work with Broadworks, another SIP based phone system. On a phone registered to Broadworks: Your phone invites the Broadworks system, Broadworks replies with a 180 (ringing) which includes a remote-party-id: field populated with the destination you are calling. That is what displays on the Polycom and Sipura 841 that I have tried. I had eneabled remote-party-id on a Cisco 7960, but something in the dialog caused the call to die. I never investigated further. --Shane Quoting "Eric \ManxPower\ Wieling" <[EMAIL PROTECTED]>: > Have you ever actually done this with Asterisk? > > Shane Young wrote: >> It would be possible if Asterisk sent a remote-party-id back to the >> calling phone. >> >> Polycom and Sipura phones (possibly Cisco phones) Support this with >> SIP on Broadworks and it works great. >> >> --Shane >> >> Quoting "Eric \ManxPower\ Wieling" <[EMAIL PROTECTED]>: >> >>> It is not possible to do this the way you want. Most phones will >>> display the called name if that name/number is in the phone's directory. --Shane ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Show Callee name on Display
It would be possible if Asterisk sent a remote-party-id back to the calling phone. Polycom and Sipura phones (possibly Cisco phones) Support this with SIP on Broadworks and it works great. --Shane Quoting "Eric \ManxPower\ Wieling" <[EMAIL PROTECTED]>: > It is not possible to do this the way you want. Most phones will > display the called name if that name/number is in the phone's directory. > > Peder @ NetworkOblivion wrote: >> We have users with Cisco 7900 phones running sip. When user A calls >> user B, we want user B's name to appear on user A's phone. It shows the >> extension they call, but not the internal name of the called user. Is >> this possible? We have some people that used to be on an MGCP based >> system and they would get the callee's name popup on their phone when >> they called someone. I can't figure out if it is possible or if it is >> just a limitation of the Cisco SIP firmware. >> >> Just to clarify with an example: >> >> 1 - Steve >> 2 - David >> >> David calls ext 1. Right now it says "calling 1". We want it to say >> "calling Steve 1". > > ___ > > Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > --Shane ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nufone problems
Quoting C F <[EMAIL PROTECTED]>: > Anybody here having any problems with nufone? > Calls are not going thru, when trying to call their customer service > number it doesn't go thru. > When trying to resolve www.nufone.net I get (sourec: > http://www.dnsstuff.com/tools/lookup.ch?name=nufone.net&type=A ): I received this when they had an outage on Wed: Date: Wed, 25 Jul 2007 17:35:07 -0400 [07/25/2007 04:35:07 PM CDT] From: "NuFone Inc." <[EMAIL PROTECTED]> Subject: Hardware Failure In Washington, DC Data Center Headers: Show All Headers At 4:36PM Eastern time we experienced a hardware failure in our Washington DC Data center. As of 5:30PM Eastern, we have restored our services at limited capacity and are working to a complete restoration of services. We apologize for the outage and are currently adding additional redundancy in our network to avoid any future outages. --Shane ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RF to IP bridge
Quoting Curt Shaffer <[EMAIL PROTECTED]>: I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is an option available for the Avaya systems but its a little out of the price range Im looking for (~$200/channel). Has anyone out there found a stable way to do this? Asterisk does this quite well: wawnmnxast1*CLI> core show application Rpt wawnmnxast1*CLI> -= Info about application 'Rpt' =- [Synopsis] Radio Repeater/Remote Base Control System [Description] Rpt(nodename[|options]): Radio Remote Link or Remote Base Link Endpoint Process. Not specifying an option puts it in normal endpoint mode (where source IP and nodename are verified). Options are as follows: X - Normal endpoint mode WITHOUT security check. Only specify this if you have checked security already (like with an IAX2 user/password or something). Rannounce-string[|timeout[|timeout-destination]] - Amateur Radio Reverse Autopatch. Caller is put on hold, and announcement (as specified by the 'announce-string') is played on radio system. Users of radio system can access autopatch, dial specified code, and pick up call. Announce-string is list of names of recordings, or "PARKED" to substitute code for un-parking, or "NODE" to substitute node number. P - Phone Control mode. This allows a regular phone user to have full control and audio access to the radio system. For the user to have DTMF control, the 'phone_functions' parameter must be specified for the node in 'rpt.conf'. An additional function (cop,6) must be listed so that PTT control is available. D - Dumb Phone Control mode. This allows a regular phone user to have full control and audio access to the radio system. In this mode, the PTT is activated for the entire length of the call. For the user to have DTMF control (not generally recomended in this mode), the 'dphone_functions' parameter must be specified for the node in 'rpt.conf'. Otherwise no DTMF control will be available to the phone user. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] zaptel huge irq problem
François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Michael L. Young > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of François Delawarde > Sent: Monday, May 14, 2007 10:24 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] zaptel huge irq problem > > Hello, > > I had noticed strange crackling sound on my phone calls going through my > zaptel device (TDM400P), so i decided to check on possible timer issue, > and found lots of issues on forums concerning the sensibility of zaptel > with IRQs, and tried about everything: moving PCI slots, noapic and > acpi=off boot options, play with different kernel options: > iosched/preemption/timer/..., play with BIOS PCI options, change > priorities, PCI latencies, IRQ balance, smp_afinity, > but impossible to come up with anything correcting that problem. > > Any idea about this? Is it possible to force the timer to ztdummy (RTC > timer) when you have a zap card plugged in? It's the only thing i could > try to make it work. > > Thanks, > François. > > Just in case: > > - Linux 2.6.18 with debian patches and xen enabled, asterisk running on > dom0. > > - Here is my zttest results under a bit of load: > # ./zttest > Opened pseudo zap interface, measuring accuracy... > 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% > 99.121094% > 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% > 99.414062% 99.902344% > 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% > 98.449707% 100.00% > > > - The card DOES NOT seem to share interrupts (checked also with lspci): > # cat /proc/interrupts >CPU0 CPU1 > 1: 1626 0Phys-irq i8042 > 6: 3 0Phys-irq floppy > 8: 0 0Phys-irq rtc > 9: 0 0Phys-irq acpi > 14: 63 0Phys-irq ide0 > 16: 1 0Phys-irq libata, eth3 > 17:6762583 0Phys-irq libata > 18: 13789 0Phys-irq libata > 19: 33459690 0Phys-irq eth1 > 20: 19864325 0Phys-irq sky2, eth0 > 21: 269250881 0Phys-irq wctdm > 256: 77735119 0 Dynamic-irq timer0 > 257:3986325 0 Dynamic-irq resched0 > 258: 37 0 Dynamic-irq callfunc0 > 259: 04652748 Dynamic-irq resched1 > 260: 0139 Dynamic-irq callfunc1 > 261: 0 28924306 Dynamic-irq timer1 > 262: 1021 0 Dynamic-irq xenbus > 263: 0 0 Dynamic-irq console > NMI: 0 0 > LOC: 0 0 > ERR: 0 > MIS: 0 > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Display Caller ID of called party
Quoting "Savoy, Kevin - Williston, ND" <[EMAIL PROTECTED]>: Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in case that matters. The polycom will display the name of the person you are calling if it receives it in the remote-party-ID. This is how it works with Broadworks. Remote-Party-ID: "Shane Young";screen=yes;party=called;privacy=off;id-type=subscriber ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linux Command Line Soft Phone - $200+ bonus
--- Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > your requirement don't really make sense. to try again: speex and/or gsm > > - put into /etc/init.d/___ - phone enabled on boot up > > Huh? IS that phone a client program? If so: why should it be run as a > server? > > There are plenty of ways to run a program at desktop startup. you are right - I'm after ease of maintenance. > > - automatically navigate around gnome and kde sound > > Huh? I have noticed in some soft phone docs that the gnome and kde sound systems need to be turned off for the soft phone to work. the phone needs to work with neither gnome nor kde running. > > - automatically navigate dhcp (if any) > > Huh? that the softphone will find its way to the server whether or not the computer is behind a router (dhcp). > > > - gnu has something sort of close(?) > > - must install through one command, thru apt-get, or thru synaptic > > On which distribution? initial install on Ubuntu 6.10 > Debian already has a host of free phones. The best seem to be Twinkle > and Ekiga for SIP and kiax for IAX. A number of others are usable. > > The only one that does *both* SIP and IAX (if you really need that) > is yate-gtk :-p . I'm after is a phone that is working when the computer boots. NO user interface - all control is done through the asterisk server to which it's connected. also, as above, not dependent on gnome or kde - with no graphics this should not be a problem. > > - must be hosted in free public place > > - must run on Ubuntu first try > > Which version? Ubuntu has some of the Debian packages. 6.10 > > phase 2: > > - have same run under Puppy Linux > > Consider giving more information on the limitations of the system > (memory? disk-space?) I'm concerned (and ignorant) about installation Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Open CallerID Database?
Our CNAM provider claims to have more than 196 million entries. I just don't think you could reliably maintain that in this format. Let's say I'm a CLEC and I have 40,000 numbers. I want to update that in one place (my SCP, probably). I wouldn't also want to update another database through another method. Quoting Robert Norton - SophMedia LLC <[EMAIL PROTECTED]>: Hey Shane, The basis of my idea was that it would be user-moderated/generated. A 'owner/operator' of a number, would submit & verify their phone number, enter their caller id, and basically be done with it. The logistics of it I don't really think would be that complicated. If a listing needs to be updated they basically go through the same process. Right now, we're using a commonly available script (I can't remember the link off hand) that uses Google, 411.com, etc, to do a lookup and although it works pretty good, it is horribly inaccurate the majority of the time. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting & Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Monday, February 19, 2007 12:46 PM To: Robert Norton - SophMedia LLC Subject: Re: [asterisk-users] Open CallerID Database? Robert On the surface, I don't see how you could a db with a very good hit rate without paying for the data. There are thousands and thousdands of database updates every day. Perhaps I am missing your intent here. Quoting Robert Norton - SophMedia LLC <[EMAIL PROTECTED]>: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting & Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. --Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Shane ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux Command Line Soft Phone - $200+ bonus
phase 1 requirements: - sip and/or iax2 using g729 and/or gsm - put into /etc/init.d/___ - phone enabled on boot up - all parameters in /etc/___ - automatically navigate around gnome and kde sound - automatically navigate dhcp (if any) - gnu has something sort of close(?) - must install through one command, thru apt-get, or thru synaptic - must be hosted in free public place - must run on Ubuntu first try phase 2: - have same run under Puppy Linux please leave bids on project here or email me directly. - Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI question
in your dialplan: [context] ... h,1,AGI(...) David Ruggles <[EMAIL PROTECTED]> wrote: I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Have a burning question? Go to Yahoo! Answers and get answers from real people who know.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone on Linux
please send me more info thanks! Tim Panton <[EMAIL PROTECTED]> wrote: On 5 Feb 2007, at 21:46, chester c young wrote: > Need to deploy between 50 to 300 lightweight Linux - only browser > and softphone. You might want to consider our lightweight java softphone (Corraleta SDK) - it can be embedded in a web page - zero install/config in the client. The UI is in HTML and javascript, so you can get it _exactly_ the way you want it. > > Any recomendations? Clearly I'm biased :-) Tim Panton www.mexuar.com www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Everyone is raving about the all-new Yahoo! Mail beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone on Linux
Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. Any recomendations? - Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problems with SJPhone (I feel stupid about this)
have a Grandstream and SJPhone SIP phones going to asterisk. with SJPhone (on Linux) getting. any ideas?? SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKc0a80264001045c3c2c52331d4920678;received=24.10.123.39;rport=60754 From: ;tag=22261807771886928353 To: ;tag=as45966c6b Call-ID: [EMAIL PROTECTED] CSeq: 41 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 - Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamically Adding A Context
In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension - The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Response on dialin - no extension
On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? Is is possible to do something like [sip-test] s,1,Answer s,2,Playback(welcome) s,3,WaitExten(30) 1,1,Noop(exten 1) ... t,1,Goto[s,2] - Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] No D-channels available! Using Primary channel16 as D-channel anyway!
> > zaptel.conf > --- > loadzone=uk > defaultzone=uk > > > span=1,1,1,ccs,hdb3,crc4,yellow > span=2,0,1,ccs,hdb3,crc4,yellow > > bchan=1-15,32-46 > dchan=16,47 > bchan=17-31,48-62 > --- > where span 1 is to the provider and span 2 is to the PBX Not sure if this matters but it sure makes it easier to read: span=1,1,1,ccs,hdb3,crc4,yellow | bchan=1-15,17-31| To provider dchan=16| span=2,0,1,ccs,hdb3,crc4,yellow | bchan=32-46,48-62 | To PBX dchan=47| >From a programmer's point of view, I would be parsing the config file looking for span and then the bchan and dchan for that span to follow. Then when another span is detected check for the bchan, dchan, fxoks, etc. to follow. I don't know if that is the case or not with zaptel drivers. > > zapata.conf > - > context=from-pstn > switchtype=dms100 > signalling=pri_cpe > callerid=asreceived > group=1 > callgroup=1 > pickupgroup=1 > rxgain=0.0 > txgain=0.0 > channel=>1-15,17-31 I would assume that you have an entry declaring a different context and group for your PBX, i.e.: context=from-pbx switchtype=dms100 signalling=pri_net group=2 callgroup=2 pickupgroup=2 rxgain=0.0 txgain=0.0 channel=>32-46,48-62 Hope this points you in the right direction. Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer - closed!
$25 to openvpn.org - thanks to Anselm Martin Hoffmeister --- Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote: > Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young: > > the answer sucks, but is apparently correct. > > If your application involves the caller (e.g. an employee of your > company) to rate the call he just did, or to enter any data to a > mysql > database over the phone right after the call, you could use the "H" > option (neither T nor h, then) and tell your phone personell about > it: > "After the call finished, press * and answer the questions the > computer > reads out to you". That way, Asterisk would (expectedly) stay in the > Audio path and even find out that the call ended if your employee did > not *g* - and your employees could cut those 7 second delays. > > Your IVR for aprés-call interaction should skip the first digit if it > happens to be an * though, because it could happen that Asterisk sees > the far end hangup just a blink before the user hits the * key. This is for volunteers calling other members of their organization, so need to keep everything low key and polite. A volunteer will call in, either by POT or SIP and will stay connected as Asterisk dials the number of the fellow member whom they've selected on a browser. The seven seconds is bad because that's a bit too long between calls - people tend to loose their concentration. Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer - closed!
the answer sucks, but is apparently correct. imho Andrew Kohlsmith is The Man, although there was someone in Germany who emailed about the T option which actually works about as well - please email me. Andrew Kohlsmith please email me. Will pay paypal if that's ok. --- Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Tuesday 16 January 2007 2:31 pm, chester c young wrote: > > however, it takes about SEVEN seconds after the called party hangs > up > > before the next priority is executed - same as with the T option. > > What kind of "last leg" are these calls? to POTS (even CAS T1) or > PRI? > > > as contrast to h option, when called party hits asterisk, the next > > priority is almost immediate. > > This is because Asterisk knows you want a hangup. > > My hunch is that you're terminating to POTS instead of PRI, and that > is how > long it takes for your telco provider to supply CPD signaling on the > analog > interface. I know Bell Canada is about that long. > > -A. > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Need Mail bonding? Go to the Yahoo! Mail Q&A for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=list&sid=396546091 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer - close
> its notransfer=yes in iax.conf not transfer=no :) this is getting close! however, it takes about SEVEN seconds after the called party hangs up before the next priority is executed - same as with the T option. as contrast to h option, when called party hits asterisk, the next priority is almost immediate. the seven second delay makes the application very difficult to use. Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
--- Paul <[EMAIL PROTECTED]> wrote: > Anselm Martin Hoffmeister wrote: > > > > Curious - is this still a $50 thread? > yes. Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
> Silly question: how are the calls going out? If they're going out > through an analog line without the ability to detect hang-ups, then, > that's the problem. > calls are coming in and out thru an Asterisk server using iax2. have tried two different DID providers and have same problem. Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial - $50 for answer
> "g" option to Dial only continues the dialplan if the destination > (called) leg of the call hangs up. It will NOT cause the dialplan to > > continue if the source (calling) leg of the call hangs up. > > When the calling channel hangs up, Asterisk will send the remaining > leg of the call to exten => "h". > this is exactly right and is exactly the problem. when the called leg hangs up the dial plan does not proceed to the next priority. Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Stumped with Dial
--- Anselm Martin Hoffmeister <[EMAIL PROTECTED]> wrote:
> Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
> > cannot make Dial(...,,g) work correctly, although Dial(...,,gh)
> works
> > just fine. (to make matters worse, it does seem to work sometimes,
> > although once working or not working between changes it either
> works or
> > doesn't work all the time.)
>
> For me,
>
> exten => 990,1,Dial(SIP/sip502,60,gro)
> exten => 990,2,Playback(special/monkeys)
> exten => 990,3,Hangup
>
> does what I would expect, and it seems reliable:
> Sip phone 502 rings, after taking the call there and hanging up, the
> caller gets the monkeys file.
>
> I use the "ro" here because it seems that makes Asterisk behave as I
> want it; "ro" should not be relevant for the "g".
>
> It does not to work though if I do not specify a timeout (60 in my
> case)...
>
> You can use the DIALSTATUS variable afterwards to find out wether a
> call
> had taken place, something like
>
> exten => 990,3,GotoIf($[X${DIALSTATUS} = XANSWER]?10)
> exten => 990,10,NoOp(Call had been answered)
>
> and do some dialplan magic :)
>
> BR
> Anselm
>
thanks, but tried gro and even Playback instead of Noop after the Dial
- still no luck. it's placing the call to a pstn thru an aix.
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[asterisk-users] Stumped with Dial - $50 for answer
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either works or
doesn't work all the time.)
extensions.conf:
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[incoming]
exten => 505111.,1,Answer
exten => 505111.,2,Noop(top)
exten => 505111.,3,Dial(IAX2/[EMAIL PROTECTED]/1501212,,g)
exten => 505111.,4,Noop(done the dial)
exten => 505111.,5,Goto(2)
after the called party hangs up nothing happens, the 4,Noop(done the
dial) is never executed.
if the dial is done with a Dial(...,,gh), and the answering phone hits
an asterisk, the next priority is executed as expected.
-- Executing [EMAIL PROTECTED]:1]
Answer("IAX2/telavoip-2", "") in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp("IAX2/telavoip-2",
"top") in new stack
-- Executing [EMAIL PROTECTED]:3] Dial("IAX2/telavoip-2",
"IAX2/telavoip/1505222||g") in new stack
-- Called telavoip/1505222
-- Call accepted by 1.2.3.4 (format ulaw)
-- Format for call is ulaw
-- IAX2/telavoip-4 is proceeding passing it to IAX2/telavoip-2
-- IAX2/telavoip-4 is ringing
-- IAX2/telavoip-4 stopped sounds
-- IAX2/telavoip-4 answered IAX2/telavoip-2
-- Channel 'IAX2/telavoip-4' ready to transfer
-- Channel 'IAX2/telavoip-2' ready to transfer
-- Releasing IAX2/telavoip-2 and IAX2/telavoip-4
-- Hungup 'IAX2/telavoip-4'
/* stays here until originating phone is hung up */
== Spawn extension (telavoip-iax-in, 1505111, 3) exited non-zero
on 'IAX2/telavoip-2'
-- Executing [EMAIL PROTECTED]:1] NoOp("IAX2/telavoip-2",
"call_loop: hungup") in new stack
-- Hungup 'IAX2/telavoip-2'
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Re: [asterisk-users] Symbolic Link
--- bilal ghayyad <[EMAIL PROTECTED]> wrote: > Hi List; > > To create the symbolic link, I read in the documenation that I have > to type this command: > > # ln -s /usr/src/'uname -r' /usr/src/linux-2.4 > > 1) What it means by 'uname -r'? > 2) Why I have to create such symbolic link to do pointing for the > kernel? For what exctly will be used with asterisk? > 3) What is the relation between creating such symbolic link and build > directory? > > Any advise. > > Regards > Bilal 1) those are backticks: `uname -r` (it prints the kernel release) 2) ? 3) don't understand your question what are you trying to do?? Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
Moises what you've done here looks great, but some examples or a little doc would be really helpful - following the bug report, although informative, is a very difficult way to extract specs. in my case I want a user to be on-line all the time - the system will dial and connect them and, when they're done, they select the next one. what I'm doing now is putting them into a loop with a g-option on the dial. the number it dials is set thru the api. if the number's not set it waits one second and loops again. from my limited knowledge using a Bridge() function is much more elegant, but I am in the dark as to what the context of the user, what happens on hangup (can it fall thru?), etc. maybe after this demo is done I'll solve this correctly using Bridge, but alas little time now for experimentations. thanks cy --- Moises Silva <[EMAIL PROTECTED]> wrote: > I have uploaded a working patch for version 1.2.12.1, and other that > seems to work in Trunk, but few people is reporting results, you can > help to get this into Asterisk, go here: > > http://bugs.digium.com/view.php?id=5841 > > The patch I ported to 1.2.12.1 is working fine, I have tested in my > servers, is the one called "bridge-1.2.12.1.patch", there are other > ones that say trunk, obviously only work with the trunk version of > Asterisk. > > Kind Regards > > On 1/3/07, chester c young <[EMAIL PROTECTED]> wrote: > > (my pstn calls are coming in thru an upstream asterisk server, so > the > > called and calling phone number is passed as an extension.) > > > > when caller comes in on 555, he will go to extension 1234 where > he > > will wait for the API to make a call to 999 for him. how do I > > bridge the two calls? > > > > extensions.conf: > > > > ;context where caller comes in > > [caller] > > 555,s,1 Answer() > > 555,s,n UserEvent(Init) ;this lets me know the connection for > > 555 > > 555,1234,1 Noop(caller waits to be bridged) > > 555,1234,2 Background(soothingmusic) > > > > ;context for connection - is this needed? > > [connect] > > > > > > from the API: > > > > (do I need to create a new context/extension first?) > > > > Action: Originate > > Channel: IAX2/upstream/999 <-- calls 999222 thru upsteam IAX > > Context: ?? > > Exten: ?? > > Priority: ?? > > > > > > > > __ > > Do You Yahoo!? > > Tired of spam? Yahoo! Mail has the best spam protection around > > http://mail.yahoo.com > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en > http://www.gnu.org"; > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] getting tones during conversation
after the Dial has connected, I want the caller (on a SIP phone) to be able to press keys in order to record call status. is this possible? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] postgres and asterisk
use a simple agi - php is easy to do. --- "O.Kamal" <[EMAIL PROTECTED]> wrote: > I just need to retrieve a value from a field in a postgres database, > and > playback this value when someone dial a specific extension. > > On 1/4/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote: > > > > O.Kamal wrote: > > > I need to retrieve my asterisk to retrieve a values from > postgresql, i > > > am looking for some sort of application like *mysql*() app, I > found one > > > but it is only available on Suse, is there any way for doing > this? > > > > > > Regards, > > > O.Youssef > > > > > What do you need to do? > > To get an SQL console with postgres you need to: > > > > psql -d -U > > > > ie: > > > > psql -d asterisk -U asterisk > > > > The location of psql is different depensing upon distribution but > > usually it's in either /usr/bin/psql or /usr/local/pgsql/bin/psql. > > > > I'm not sure if this is what you want, if you want a pretty GUI > > front-end then you could look at Pgadmin III (www.pgadmin.org) > which > > will run on Windows 2000/XP/2003 or unix/linux running X and > requires > > wxWindows and a pile of common libraries. > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
how is this fitting into 1.4? - can it be compiled against 1.4 or only 1.2? - if not, are there leanings in that direction? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ztdummy on 1.6
does anyone know if ztdummy is requires under 1.6 or are they using Linux' rtc? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] API: how to bridge originated call?
Moises this sounds great! three questions if you don't mind: 1. how is this fitting into 1.6? 2. are there some directions I can follow for downloading the right source and applying your patches? 3. is there a central place for doc on your patches? (if not I would be more than happy to write it) thanks cy --- Moises Silva <[EMAIL PROTECTED]> wrote: > By the way, Chester, please report results to the bug I sent you, is > very imortant the users feedback to get this into Asterisk > > Regards > > On 1/3/07, Moises Silva <[EMAIL PROTECTED]> wrote: > > I have uploaded a working patch for version 1.2.12.1, and other > that > > seems to work in Trunk, but few people is reporting results, you > can > > help to get this into Asterisk, go here: > > > > http://bugs.digium.com/view.php?id=5841 > > > > The patch I ported to 1.2.12.1 is working fine, I have tested in my > > servers, is the one called "bridge-1.2.12.1.patch", there are other > > ones that say trunk, obviously only work with the trunk version of > > Asterisk. > > > > Kind Regards > > > > On 1/3/07, chester c young <[EMAIL PROTECTED]> wrote: > > > (my pstn calls are coming in thru an upstream asterisk server, so > the > > > called and calling phone number is passed as an extension.) > > > > > > when caller comes in on 555, he will go to extension 1234 > where he > > > will wait for the API to make a call to 999 for him. how do > I > > > bridge the two calls? > > > > > > extensions.conf: > > > > > > ;context where caller comes in > > > [caller] > > > 555,s,1 Answer() > > > 555,s,n UserEvent(Init) ;this lets me know the connection for > > > 555 > > > 555,1234,1 Noop(caller waits to be bridged) > > > 555,1234,2 Background(soothingmusic) > > > > > > ;context for connection - is this needed? > > > [connect] > > > > > > > > > from the API: > > > > > > (do I need to create a new context/extension first?) > > > > > > Action: Originate > > > Channel: IAX2/upstream/999 <-- calls 999222 thru upsteam IAX > > > Context: ?? > > > Exten: ?? > > > Priority: ?? > > > > > > > > > > > > __ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam protection around > > > http://mail.yahoo.com > > > ___ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > -- > > "Su nombre es GNU/Linux, no solamente Linux, mas info en > http://www.gnu.org"; > > > > > -- > "Su nombre es GNU/Linux, no solamente Linux, mas info en > http://www.gnu.org"; > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] API: how to bridge originated call?
(my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 <-- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sangoma A102 w/ EC module gets intermittent echo/audio artifacts
< Zaptel.conf: < loadzone = us < span=1,1,0,esf,b8zs < span=2,1,0,esf,b8zs < bchan=1-23 < dchan=24 < bchan=25-47 < dchan=48 Just a quick thought in looking at the settings above, it appears that you have set both spans as the primary timing source. I am pretty sure that only one span should be the primary timing source. The other span should either be at 0 (not used as a timing source) or set as a secondary timing source. Hope this helps. Michael L. Young ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Hardware for Asterisk Server?
--- Mark Greene <[EMAIL PROTECTED]> wrote: > Hey guys, > > In your experience what is the best way to go for a production > asterisk box in your offices? (In the US) I have had very good luck with Opterons in Tyson rackmounts bought from Newegg. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial - g option
Dial(...|30|g) does not seem to work whereas Dial(...|30|gh) works just fine __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
