Re: [asterisk-users] Hints no longer working in 1.4beta3 with Polycoms
Hints are not working in 1.4b3 period. Snom 360s do not show any status updates. However, before you file a bug report you might want to check to see if there are changes to the way hints are implemented in 1.4. It might be a configuration problem rather than a bug but I have not had time to look into it. John On Tue, 2006-11-21 at 16:54 -0800, Anthony Rodgers wrote: Hi there, Is there anyone else using hints and buddy watch on 1.4beta3 with Polycoms? If so, can you give an indication of whether they are working or not? We had hints working fine on 1.2.1, but they have stopped working after upgrading our test server to 1.4beta3. We've tried rebooting the phones, 'sip reload', deleting and recreating the directory entries etc. A 'sip debug' shows absolutely no NOTIFY or XML presence messages as calls progress.. Next stop Mantis :-) CP ___ --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
[Asterisk-Users] The CAVP is now accepting memberships applications
On-line signup form are available on our website at www.cavp.ca in the Membership section or please call 1-866-940-CAVP (2287) and select option 3 (CAVP treasurer). -- The CAVP is now accepting memberships applications. This is a pivotal moment for the CAVP and we need your support. Establishing a solid membership base will determine the organizations ability to function as an effective association. CAVP Membership is available to any interested party who is involved in the Voice Over IP industry. Further details including a link to our on-line signup form are available on our website at www.cavp.ca in the Membership section. Founding members will be given special recognition within the CAVP and on the CAVP web site including prominent links to your business. To be considered a founder, members must join soon and also pay a one time membership fee which is twice their normal yearly fee. Benefits for Members Membership in CAVP provides many benefits which can have a direct impact on your profitability. CAVP Members: - Have the right to display the CAVP logo on the web sites and any of their business material. - Are eligible to vote at CAVP meetings for board elections and general policy directions. - Have representation at CRTC working group meetings for issues such as E911, ENUM, and network interconnection. - Can ask the CAVP for legal advice with regard to regulatory issues and compliance. - Granted access to members only area of the web site containing complete listings of members contact information as well as documents as listed below. - Allowed to receive and post to the CAVP mailing lists. - Can utilize the CAVP's library of legal documents and disclaimers for customers to ensure they are in compliance with CRTC regulations. - Entitled to use the CAVP's ENUM system for direct peering with other CAVP members. - Participation in the CAVP IP Gateway service which allows members to terminate calls through other members PSTN gateways in each region. Disclaimer: As this organization is newly formed not all of the member benefits are fully in place at this time. -- Regards, John Lange Canadian Association of VoIP Providers 1-866-940-CAVP (2287) ext 2. ___ --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] Changes to sip.conf in 1.2.x ?
Is there something significantly different in 1.2.x sip.conf that would prevent clients from registering with the server? We installed 1.2.3 and copied over sip.conf from a production 1.0.9 box and clients can not register. Asterisk just replies with unauthorized. Here is a sample from sip.conf. [2048881234x3] accountcode=318 type=friend context=openit username=2048881234x3 callerid=OpenIT 2048881234 secret=xx canreinvite=no host=dynamic mailbox=2048881234 nat=yes qualify=yes dtmfmode=rfc2833 disallow=all allow=g726 allow=gsm allow=ulaw allow=ilbc allow=g729 Regards, -- John Lange ___ --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] Stopping Asterisk from forwarding calls?
Ultimately this turned out to be a red herring as well. dialparties.agi just does a database dip to figure out which extensions are forwarded and then builds a dialstring based on whats left. It then returns to the Asterisk dialplan and the extensions are still dialed in the normal way. I stopped looking at this point but it appears that it only works if you are using AMP to manage your extensions in a database. Unless someone has a better idea it looks like the only way to do this will be a patch to Asterisk. Thanks all for your suggestions. -- John Lange On Mon, 2005-11-07 at 23:20 -0600, John Lange wrote: Thanks Tad. This might turn out the be the clue I was looking for. It appears AMP has a macro-dial which has a comment about dealing with CFWD, DND etc. It actually dials using a script: exten = s,4,AGI,dialparties.agi I'm still trying to figure out what it does exactly because the code is not commented very well but it looks promising. Thanks for pointing me in this direction. John On Mon, 2005-11-07 at 15:35 -0500, Tad Heckaman wrote: I use [EMAIL PROTECTED], and if I have one of my Cisco phones forwarded to my cell phone, when the phones ring in a ring group, it never forwards. You may want to look at the latest configs that comes with [EMAIL PROTECTED] and see if theres some special dialplans thats doing what your looking for. Keep in mind I am using the call forward on the phone, and not the built in call forward in the dialplan. On 11/7/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: John Lange wrote: Reading the source code I see there are two parameters for channels, allowredir_in allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring?). Those are entirely unrelated. At this time there is no method available to make Asterisk ignore incoming '302 REDIRECT' from SIP phones. It may be possible to send those 'forward' requests to a context that has no valid extensions in it, but I don't think we even support that at this time. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stopping Asterisk from forwarding calls?
The first time I asked this to the list I didn't do a great job of it so I'm posting again with more details. Problem: when ringing multiple extensions, if one user has their phone forwarded directly to voicemail, it stops the whole group from ringing because the voicemail picks up immediately. Also, after hours incoming calls are to ring all extensions so anyone can pickup. But if one person in the office has their phone forwarded the same problem occurs. What we need is for asterisk, when ringing multiple extensions, to completely ignore the forward requests and just ring the remaining phones. Reading the source code I see there are two parameters for channels, allowredir_in allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring?). -- John Lange ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?
Thanks Tad. This might turn out the be the clue I was looking for. It appears AMP has a macro-dial which has a comment about dealing with CFWD, DND etc. It actually dials using a script: exten = s,4,AGI,dialparties.agi I'm still trying to figure out what it does exactly because the code is not commented very well but it looks promising. Thanks for pointing me in this direction. John On Mon, 2005-11-07 at 15:35 -0500, Tad Heckaman wrote: I use [EMAIL PROTECTED], and if I have one of my Cisco phones forwarded to my cell phone, when the phones ring in a ring group, it never forwards. You may want to look at the latest configs that comes with [EMAIL PROTECTED] and see if theres some special dialplans thats doing what your looking for. Keep in mind I am using the call forward on the phone, and not the built in call forward in the dialplan. On 11/7/05, Kevin P. Fleming [EMAIL PROTECTED] wrote: John Lange wrote: Reading the source code I see there are two parameters for channels, allowredir_in allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring?). Those are entirely unrelated. At this time there is no method available to make Asterisk ignore incoming '302 REDIRECT' from SIP phones. It may be possible to send those 'forward' requests to a context that has no valid extensions in it, but I don't think we even support that at this time. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tad Heckaman ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem ringing multiple extensions when one is forwarded
We have an incoming line which rings a large group of phones. If one of the phones is set to call-forward, the entire group is diverted. We would like asterisk to ignore the forward and continue to ring the rest of the phones. Any ideas how this could be done? I suspect that ring groups could be used to solve this problem but the documentation is very light in this area. By the way, the phones are Cisco 7912s 7940s and the forwarding is set on the phones themselves, not in asterisk. Regards, -- John Lange ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem ringing multiple extensions when one is forwarded
Here is a bit more information. First, to clarify, you are correct the entire group is not diverted, however, since the forward is going to a direct to voicemail extension it answers immediately and that stops the group from ringing. What we need is for asterisk to completely ignore the forward and just ring the remaining phones. BTW, pri_gw below is a Cisco sip gateway connected to a PRI. Here is a sanitized mini-version of the CLI output. -- Executing Dial(SIP/10.0.0.1-b5437768,SIP/EXTEN1SIP/EXTEN2SIP/EXTEN3.. etc. -- Called EXTEN1 -- Called EXTEN2 -- Called EXTEN3 -- SIP/EXTEN1-848c is ringing -- SIP/EXTEN2-8d15 is ringing -- Got SIP response 302 Moved Temporarily back from 10.0.0.56 -- Now forwarding SIP/10.0.0.1-b5437768 to 'Local/forwardnum@context' (thanks to SIP/EXTEN3-7642) -- Executing SetCallerID(Local/forwardnum@context-81d5,2, Name 2021234567) in new stack -- Executing Dial(Local/forwardnum@context-81d5,2, SIP/forwardnum@pri_gw) in new stack -- Called forwardnum@pri_gw -- SIP/pri_gw-6f36 is making progress passing it to Local/forwardnum@context-81d5,2 -- Local/forwardnum@context-81d5,1 is making progress passing it to SIP/10.0.0.1-b5437768 -- SIP/pri_gw-6f36 answered Local/forwardnum@context-81d5,2 -- Local/forwardnum@context-81d5,1 answered SIP/10.0.0.1-b5437768 -- Attempting native bridge of SIP/10.0.0.1-b5437768 and SIP/pri_gw-6f36 Thanks, John On Sun, 2005-11-06 at 14:23 -0500, C F wrote: I'm not so sure that the entire group is diverted. Lets see first: 1. How are you calling these phones? 2. Are you using Zap? 3. If the forward is to a local extensions, does the same thing happen? Also please post your CLI output. For some reason I think you are using Zap channels, and the Cisco phone is forwarded to an external number that uses a Zap FXO port, which to asterisk is answered as soon as it starts dialing, the workaround might be to put a c in the dial coommand, which requires a confirmation when the phone rings to be considered answred. Or you could simply block phone enabled forwards that involve using Zap FXO ports. On 11/6/05, John Lange [EMAIL PROTECTED] wrote: We have an incoming line which rings a large group of phones. If one of the phones is set to call-forward, the entire group is diverted. We would like asterisk to ignore the forward and continue to ring the rest of the phones. Any ideas how this could be done? I suspect that ring groups could be used to solve this problem but the documentation is very light in this area. By the way, the phones are Cisco 7912s 7940s and the forwarding is set on the phones themselves, not in asterisk. Regards, -- John Lange ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Canadian Association of VoIP Providers
My apologies for the cross-posting. If you are a business or individual providing Voice over IP services in Canada then we encourage you to read this email carefully otherwise please disregard. - As you are most likely aware, the CRTC has undertaken the roll of regulating VoIP services in Canada and is currently conducting hearings with the goal of putting in place regulatory requirements for all VoIP providers. Specifically, the CRTC's CISC VoIP 911 working group ( http://www.crtc.gc.ca/cisc/eng/cisf3e4_20.htm ) is very actively looking at what regulations to put in place in order to implement E911 services for VoIP. The recommendations of this committee will have direct impact on your business. Currently this working group is is largely comprised by the Local Exchange Carriers (ILECs CLECs) with representation from the large VoIP providers (Primus Vonage). To date only a very few smaller VoIP providers are participating. Subsequently, much of the discussion is oriented around solutions designed to work in the traditional telco world. Depending on your companies infrastructure these solutions may be very expensive or completely impossible for your business to implement. Some members of the working group are even of the position that VoIP service be abolished altogether. Your companies direct participation in the hearings is the best way to have an impact. However, we acknowledge that not all companies have the time and/or resources to fully participate lengthy public hearings. It is with this in mind we propose the formation of a Canadian industry association for VoIP providers and we invite you to participate. The short term goal is to contact and organize Canadian VoIP providers into a formal association. Longer term the association will work towards the following goals: - Keep VoIP providers informed about current regulatory issues - Ensuring VoIP providers have a place at the CRTC table - Develop industry recommendations - Communicate industry recommendations to the CRTC working group - Communicate industry positions to the media - Other (to be determined by the association) At the outset it is envisioned that this group would work in the following way: - No membership fee - Regular updates via email list - Frequent Conference calls - No face-to-face meetings (no travel) - Development of an Industry web site - In-person representation at each CRTC meeting (The CRTC working group meets monthly in a different province each month. We hope to have at least one member representative attend each meeting.) To voice your support (or opposition) for the formation of this group please contact me directly either by email or telephone (contact information in the signature). It is important that you do not delay. CISC working group recommendations to the CRTC are forthcoming. You will be contacted with details on how to participate in the formation of this association. Our intention is to hold our first conference call as early as possible (early next week). NOTE: No web site or association material yet exists because the group has not been officially formed and named. This will be one of the first items of business for the new group. Regards, -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playtones volume control?
We have a script which lets the phone ring 3 times and then answers it to listen for a fax tone. While listening for the fax tone it plays a ring with the simple: exten = s,4,Playtones(ring) The problem is, the ring is about 10 times louder than the previous 3 rings. Is there a way to control the volume on Playtones or perhaps a better way of implementing this? -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P
No, there is nothing else connected to the lines. Also got a single report of a call coming in, but when answered hearing a ringing sound on the line (as if you were placing an outbound call). Incoming caller doesn't hear the ringing but hears the person say hello. This is very strange. -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location On Thu, 2005-06-09 at 19:10 -0700, Steve Totaro wrote: - Original Message - From: John Lange [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 09, 2005 1:26 PM Subject: [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P We have a client that has a single Wildcard TDM400P with 3 FXO ports on Asterisk 1.0.7. Occasionally the system seems to loose its mind and starts originating calls from that Zap channels that don't exist. The receptionist picks up the phone and nobody is there. This can happen repeatedly over and over again within a few minutes. As far as we can tell these are definitely not real calls as nobody has ever called back and said they couldn't get through. Does anyone have a suggestion for why this might be happening? -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location Is there an alarm system, fax or any other sharing the line? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P
We have a client that has a single Wildcard TDM400P with 3 FXO ports on Asterisk 1.0.7. Occasionally the system seems to loose its mind and starts originating calls from that Zap channels that don't exist. The receptionist picks up the phone and nobody is there. This can happen repeatedly over and over again within a few minutes. As far as we can tell these are definitely not real calls as nobody has ever called back and said they couldn't get through. Does anyone have a suggestion for why this might be happening? -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Authentication problem between Cisco router and Asterisk when calls are forwarded
We are using a Cisco router with a T1 card plugged into a PRI provided by a local telco (Allstream). This Cisco accepts calls and sends them to a couple of servers running Asterisk depending on which number was dialled. But there is a problem. When a call comes in to the Cisco from the PSTN it sends it to the Asterisk server something like this: FROM: 204XXX@CISCO IP TO: 204NNN@Asterisk IP Normally, this is no problem. The user 204XXX does not exist on the Asterisk server because it is the callerid of someone on the PSTN. However, if a number on the PSTN is forwarded to a number on the Asterisk server, and then someone else on the Asterisk server calls the PSTN number, the call appears at the Asterisk server as being from a local caller and it is rejected because it has no username/password. I know, its confusing. So let me try and simplify. Lets say 204 791 2345 is my cell phone. And 204 885 0872 is my office phone. When I get into the office, I forward my cell to my office phone to save airtime. So 204 791 2345 is forwarded to 204 885 0872. A random outside caller (204 123 4567) phones my cell (204 791 2345), which is forwarded to 204 885 0872. No problem, the calls appears at the Asterisk server as FROM: 2041234567@CISCO IP. Since 2041234567 is not a user on the Asterisk system it falls through to the default context and no username/password is required. However, if someone on a VoIP phone (lets say 204 444 ) connected to the Asterisk server calls my cell, the Asterisk server rightly believes the call is destined for the PSTN and routes it to the Cisco which sends it out to the PSTN where it promptly comes back in the PRI (because of the forwarding) and is returned back to the Asterisk box. The problem is, the from is now FROM: 20@CISCO IP, and 20 *IS* a valid user on the Asterisk box so Asterisk tries to authenticate the user. The Cisco of course knows nothing about the username/password for that user and the call gets rejected. I am not a Cisco person; so the question is, is it possible to have one of the following: 1) Have asterisk lock onto the IP address of the FROM instead of the userid portion? 2) Have the Cisco authenticate (register) as a SIP client to the Asterisk server. This allows me to place the Cisco in its own context. 3) Have the Cisco override the FROM portion inserting its own information but still passing the correct callerID information? 4) another suggestion? Thanks, -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [cisco-voip] SIP Authentication problem between Cisco router and Asterisk when calls are forwarded
On Tue, 2005-05-31 at 13:38 -0400, Jared Mauch wrote: On Tue, May 31, 2005 at 12:30:07PM -0500, John Lange wrote: I am not a Cisco person; so the question is, is it possible to have one of the following: 1) Have the Cisco authenticate (register) as a SIP client to the Asterisk server. This allows me to place the Cisco in its own context. Nope, see below tho.. 2) Have the Cisco override the FROM portion inserting its own information but still passing the correct callerID information? 3) Some other solution that I'm not thinking of? perhaps something like this: [pstnlink] type=friend host=1.2.3.4 nat=no qualify=3000 context=long-distance-capable-context insecure=yes insecure=very Thanks for that suggestion. I have tried that in the past and it does not work. Unfortunately, Asterisk only falls back to using the IP address for determining context if no matching username is found. Since we always have a matching username it ignores the IP address even though that client is already registered from another IP. Very frustrating... I'm starting to think the only solution would be to hack the Asterisk source code so it prefers IP address over username when host=an ip address. However I'm very reluctant to do this unless it ends up merged into the main code for obvious reasons. -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interrupting voicemail with *, dropping toa extension. Does it work?
Well I finally got it working but the solution doesn't make sense to me.
If you send a user to voicemail as such:
[stdexten]
exten = 1234,1,Dial(1234,20)
exten = 1234,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u1234)
When the user presses * (or 0 if you have that set), they are returned
in the context voicemail, NOT [stdexten] ! The call is NEVER in the
context voicemail so why would it return there? All I can think of is
it conflicts with the command Voicemail in some way...
So to make * work you must have:
[voicemail]
exten = a,1,VoiceMailMain()
Does this make sense to anyone?
--
John Lange
President OpenIT ltd. www.Open-IT.ca (204) 885 0872
VoIP, Web services, Linux Consulting, Server Co-Location
On Thu, 2005-05-12 at 20:09 -0700, Jim Sturtevant wrote:
You should set operator=yes in voicemail.conf to get 0 out to work.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Lange
Sent: Thursday, May 12, 2005 7:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Interrupting voicemail with *, dropping
toa extension. Does it work?
On Fri, 2005-05-13 at 12:25 +1000, Adam Goryachev wrote:
On Thu, 2005-05-12 at 20:44 -0500, John Lange wrote:
Very very odd.
Its not a DTMF problem because other tones work fine. # for example
skips the OGM as it should.
So could it possible be a config issue?
The voicemail box in question is in the [default] context inside
voicemail.conf.
[default]
2048850872 = ,John Lange,[EMAIL PROTECTED]
That tells me I need a this in extensions.conf:
Does it??
[default]
exten = a,1,VoicemailMain() ; If they press *, send the user into
VoicemailMain
Am I missing something blazingly obvious?
I thought that it should be like this:
[somecontext]
exten = 1234,1,Voicemail([EMAIL PROTECTED]) --- Standard check VM
exten = a,1,VoiceMailMain() --- press * to get here
ie, the a extension should be in the same context as the voicemail
extension, the voicemail context (default) is irrelevant in all this...
It is ONLY used internally by the voicemail app to determine which
mailbox this is. Don't confuse them just because they are both called
context's.
I created this bare-bones example to test it.
[mycontext]
exten = 8761234,1,Voicemail(u2048761234)
exten = a,1,VoiceMailMain()
It does not work. As mentioned I can skip the OGM by pressing #, but *
(and 0) do nothing.
With verbose set to 9 I see nothing on the console for any of the key
presses.
By the way, I'm using a recent CVS version of Asterisk.
Asterisk CVS-HEAD-05/03/05-16:21:27
This is very baffling.
Are there any other ways of trouble shooting it?
I didn't think it could be a config issue because I thought it should at
least show the * in the console and then complain about no a
extension or something but I get absolutely nothing in the console.
Dunno, but try the above... and let us know.
Regards,
Adam
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[Asterisk-Users] Interrupting voicemail with *, dropping to a extension. Does it work?
I've played around with the lightly documented Asterisk voicemail feature whereby a caller can press * during the playback of the OGM and be returned to the a extension in the context of the voicemail box. No matter what, Asterisk does nothing when you press *. It does not interrupt the OGM and it certainly does not return to the a context. Watching the console there is no indication of any key press. Has anyone ever got this working? Is there an undocumented setting in asterisk someplace which enables this feature? -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupting voicemail with *, dropping to a extension. Does it work?
Very very odd. Its not a DTMF problem because other tones work fine. # for example skips the OGM as it should. So could it possible be a config issue? The voicemail box in question is in the [default] context inside voicemail.conf. [default] 2048850872 = ,John Lange,[EMAIL PROTECTED] That tells me I need a this in extensions.conf: [default] exten = a,1,VoicemailMain() ; If they press *, send the user into VoicemailMain Am I missing something blazingly obvious? I didn't think it could be a config issue because I thought it should at least show the * in the console and then complain about no a extension or something but I get absolutely nothing in the console. Thanks, -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location On Thu, 2005-05-12 at 20:30 -0400, John covici wrote: Works fine here. on Friday 05/13/2005 Ronald Wiplinger([EMAIL PROTECTED]) wrote John Lange wrote: I've played around with the lightly documented Asterisk voicemail feature whereby a caller can press * during the playback of the OGM and be returned to the a extension in the context of the voicemail box. No matter what, Asterisk does nothing when you press *. It does not interrupt the OGM and it certainly does not return to the a context. Watching the console there is no indication of any key press. Has anyone ever got this working? Is there an undocumented setting in asterisk someplace which enables this feature? Have you tried to set dtmf to rfc2833??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupting voicemail with *, dropping to a extension. Does it work?
On Fri, 2005-05-13 at 12:25 +1000, Adam Goryachev wrote: On Thu, 2005-05-12 at 20:44 -0500, John Lange wrote: Very very odd. Its not a DTMF problem because other tones work fine. # for example skips the OGM as it should. So could it possible be a config issue? The voicemail box in question is in the [default] context inside voicemail.conf. [default] 2048850872 = ,John Lange,[EMAIL PROTECTED] That tells me I need a this in extensions.conf: Does it?? [default] exten = a,1,VoicemailMain() ; If they press *, send the user into VoicemailMain Am I missing something blazingly obvious? I thought that it should be like this: [somecontext] exten = 1234,1,Voicemail([EMAIL PROTECTED]) --- Standard check VM exten = a,1,VoiceMailMain() --- press * to get here ie, the a extension should be in the same context as the voicemail extension, the voicemail context (default) is irrelevant in all this... It is ONLY used internally by the voicemail app to determine which mailbox this is. Don't confuse them just because they are both called context's. I created this bare-bones example to test it. [mycontext] exten = 8761234,1,Voicemail(u2048761234) exten = a,1,VoiceMailMain() It does not work. As mentioned I can skip the OGM by pressing #, but * (and 0) do nothing. With verbose set to 9 I see nothing on the console for any of the key presses. By the way, I'm using a recent CVS version of Asterisk. Asterisk CVS-HEAD-05/03/05-16:21:27 This is very baffling. Are there any other ways of trouble shooting it? I didn't think it could be a config issue because I thought it should at least show the * in the console and then complain about no a extension or something but I get absolutely nothing in the console. Dunno, but try the above... and let us know. Regards, Adam -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why is host= being ignored in sip.conf ?
Perhaps someone can help me with a problem that has me thoroughly stumped! Given the below sip.conf file, why do calls that come in from server 123.456.789.012 NOT go into the ext context? I've tried many variations of the below but no matter what I try calls from that server are always in the default context. Is it possible that Asterisk must first register to the server before it will accept the host context? In this case the server in question is a Cisco router hooked up to a PRI. It does not need or allow registrations so I hope that is not the problem. [general] context=default bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=g729 allow=ulaw allow=alaw [gw] type=peer context=ext host=123.456.789.012 canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw -- John Lange ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7905/7912, SIP, g729 and DTMF setup
If anyone sees any mistakes in the following advice, please let me know. I recently went through a bit of a configuration nightmare with the Cisco 7905 phone using the g729 codec and Asterisk and I thought I share it here for anyone who might be searching for help on this in the future. The setup is three, 7905 Cisco phones with the SIP firmware attached to a Asterisk server remotely through a NAT firewall. The Asterisk is connecting to the PSTN via a SIP gateway (a Mediatrix box) which only uses g711. This setup was actually fairly easy but the one nagging problem was the DTMF tones. We tried numerous configurations with different results. Sometimes DTMF tones would work on outbound calls but not on the Asterisk voice mail system. Other times they would work for voicemail but no tone would be heard on the outside call. Even more frustrating, sometimes we could get DTMF if the call was placed outbound, but incoming calls had no DTMF. Anyhow, here is what I learned. 1. When using a Cisco phone with the g729 codec, your sip.conf should be as follows (simplified): [XXX] type=friend context=local username=XXX callerid=XXX secret=XXX host=dynamic mailbox=XXX nat=yes qualify=yes dtmfmode=rfc2833 ; * See note. canreinvite=no disallow=all allow=g729 * Note: If you use g729 you can not use inband. Documentation on the voip-wiki seems to indicate that you should use dtmfmode=info with the Cisco phone but I found this does NOT work end-to-end with outbound, inbound, and voicemail system. The settings on the Cisco phone are also very important. They should be: RxCodec:3 ;g729 TxCodec:3 ;g729 AudioMode :0x0020 ; DTMF signalling Always out-of-band * Note: remember you have to buy g729 licenses for Asterisk from digium. On the flip side, the gateway is set as follows: [mediatrix] type=peer context=mediatrix host=xxx.xxx.xxx.xxx dtmfmode=inband ; inband works with g711 only disallow=all allow=ulaw allow=alaw I hope this helps someone. -- John Lange ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 302 Moved Temporarily Do not disturb
On Sat, 2004-12-18 at 15:07, Eric Wieling aka ManxPower wrote:
John Lange wrote:
I have some Cisco 7905 phones with the SIP load 1.02.00(040406A).
When the phone is off-hook but no call has been placed, or when the Do
Not Disturb is activated, the phone returns a 302 Moved Temporarily
message back to asterisk as follows:
Third, (if not), are there any work arounds or suggestions for this?
;
; Give Voicemail extension XX09
;
exten = _XX09,1,GoToIf($[X${RDNIS} != X]?${EXTEN},4)
exten = _XX09,2,VoicemailMain()
exten = _XX09,3,Hangup
exten = _XX09,4,VoiceMail(u${RDNIS})
exten = _XX09,5,Hangup
Ok, it took me a while to figure out what this was doing but let me say
this is a thing of beauty and it works perfectly.
Here is the version I ultimately implemented with a couple of comments.
-
; give voicemail at the traditional 8500
[voicemail]
exten = 8500,1,GoToIf($[X${RDNIS} != X]?${EXTEN},4) ; this allows call forward
to voice mail and therefore fixes the Cisco DND problem.
exten = 8500,2,VoicemailMain(s${CALLERIDNUM}) ; by passing callerid it takes
us directly to our own mailbox
exten = 8500,3,Hangup
exten = 8500,4,VoiceMail(u${RDNIS}) ; passing the original dialed number gives
us the correct mailbox.
exten = 8500,5,Hangup
-
Thanks greatly to Eric Wieling!
--
John Lange
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[Asterisk-Users] Problem with 302 Moved Temporarily Do not disturb
I have some Cisco 7905 phones with the SIP load 1.02.00(040406A). When the phone is off-hook but no call has been placed, or when the Do Not Disturb is activated, the phone returns a 302 Moved Temporarily message back to asterisk as follows: --- -- Executing Dial(SIP/5060-0811bb00, SIP/9871234|20|Ttr) in new stack -- Called 9871234 -- Got SIP response 302 Moved Temporarily back from 24.xx.xxx.6 -- Now forwarding SIP/5060-0811bb00 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/9871234-b2ca) --- This forwarding doesn't work because the Voice mail system does not know which extension was originally dialed so I either get a please enter the mail box number or, if it is a local extension originating the call, a You have no new messages message. First, I need to know if it is the Cisco phone, or Asterisk that is automatically doing the forwarding? It certainly is not something I setup in the dialplan. Second, are their any ways to trap a 302 message in the dialplan so it does the right thing? Third, (if not), are there any work arounds or suggestions for this? Thanks, -- John Lange ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Cisco 7905 Not Acceptable Here
Asterisk seems to have a problem with the Cisco 7905. If the user is on the phone with another call, asterisk reports: -- Executing Dial(SIP/206.XX.XXX.XXX-08f899d8, SIP/204X83|10) in new stack -- Called 2044X83 -- Got SIP response 488 Not Acceptable Here back from 192.168.1.112 And it then rolls as a NOANSWER rather than a BUSY. If you just take the phone off the hook but don't place a call, it then reports a BUSY as it should. Is this an Asterisk bug or a Cisco configuration problem? If it is related to call waiting can this be disabled? Thanks, -- John Lange ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with Cisco 7905 Not Acceptable Here
Thanks that suggestion has shed some light on the problem. I have only the g729 codec allowed and I have the licenses and the codec installed on asterisk. Calling works fine in, out, and between extensions so the codec is setup fine. The only time I get that error is if I call an extension that is in use. I assume this is normally the time some sort of callwaiting sound would play. As a test I disabled the g729 codec and enabled only ulaw and the problem went away so it is clearly a problem with the 729 codec and the call waiting sound. Now that I have isolated the problem I will post a new thread. Thanks. John Lange On Mon, 2004-11-01 at 11:23, Eric Wieling wrote: John Lange wrote: Asterisk seems to have a problem with the Cisco 7905. If the user is on the phone with another call, asterisk reports: -- Executing Dial(SIP/206.XX.XXX.XXX-08f899d8, SIP/204X83|10) in new stack -- Called 2044X83 -- Got SIP response 488 Not Acceptable Here back from 192.168.1.112 You prolly have allow=all in sip.conf which will allow all sorts of codecs that are not supported by the Cisco. For testing do disallow=all and allow=ulaw. If that works you can try allow'ing the codec you want instead of ulaw. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John Lange ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call waiting does not work with g729 codec
I have purchased the g729 codec from Digium and I'm using it in conjunction with Cisco 7905 phones. When a call is placed to an extension which is in use, instead of sounding the call waiting tone, it causes an error: -- Executing Dial(SIP/206.XX.XXX.XXX-08f899d8, SIP/204X83|10) in new stack -- Called 2044X83 -- Got SIP response 488 Not Acceptable Here back from 192.168.1.112 If I change the codec to ulaw the problem goes away and the tone is played normally. There are two issues here: 1) The call waiting codec problem. 2) The error causes asterisk to return a NOANSWER status when it should return something else. More sensible would be BUSY or UNAVAILABLE. This is significantly important for me because in this case, when someone is on the phone the dialplan is supposed to roll-over to the next available extension. Only when it exhausts all extensions or when it gets a NOANSWER should it go to voice mail. In short, BUSY should roll-over. NOANSWER should go to voice mail. Since the error is returning a NOANSWER status code I can't setup my dialplan properly. And one final question, is it possible to disable call waiting? -- John Lange ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is there a way to disable call wating?
I would like to completely disable call waiting. Does Asterisk have an option for that? Thanks, -- John Lange ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ringing
Just a note to the list, I tried to apply that patch to the current CVS but it failed. This was expected of course because the code has changed since the patch was released. Assuming this patch is stable it should defiantly be Incorporated into the main code. John Lange On Tue, 2003-03-04 at 23:18, TC wrote: on a zap channel you can do for example exten - 1,2,Dial(Zap/1rN) where rN is (r1=quick chip+normal ring; r2=British style ringing; r3=three short bursts; r4=long ring). see also this patch to allow userdefined ring candence http://www.marko.net/asterisk/archives/0212/0318.html maybe CAM has kept it current could pass along for inclusion in cvs base -Original Message- From: Brian Johnson [EMAIL PROTECTED] To: [EMAIL PROTECTED] [EMAIL PROTECTED] Date: March 4, 2003 7:27 PM Subject: Re: [Asterisk-Users] Distinctive ringing As a sidenote ... can asterisk generate distinctive ringing for the analog extensions? Jim Archer ([EMAIL PROTECTED]) wrote*: Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- John Lange [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
