Re: [asterisk-users] RTP traffic through Asterisk??
Can you move the transfer functionality to the end device rather than through Asterisk? That's what we do - John On Tue, 2009-11-17 at 14:07 +0100, Ignacio wrote: Thank you very much to both of you. My problem was that I used transfer in the dialplan. I have read that If I have Tt, wW, or hH, then asterisk will always stay in the path. So I have to redefine what I want to do know. Allowing transfers is an useful feature, but I wanted all rtp traffic went p2p. Is there any intermediate solution? Thanks. Regards Ignacio On Mon, Nov 16, 2009 at 7:52 AM, Leonja Cerebro lio...@gmail.com wrote: see the DTMF method on both phones. 2009/11/14 Ignacio sanfermi...@gmail.com Ok, thank you very much. I will try to find any information in asterisk documentation about RTP. On Fri, Nov 13, 2009 at 3:03 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? snip I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 ___ -- 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 -- We never did too much talking anyway So don't think twice, it's all right -- There are more things in heaven and earth, Horatio, Than are dreamt of in your philosophy. ___ -- 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 ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] RTP traffic through Asterisk??
On Fri, 2009-11-13 at 11:44 +0100, Ignacio wrote: I have just established a call between 2 sip phones and I have noticed that all RTP traffic goes through Asterisk Server. I was expecting RTP traffic went to one phone to another phone directly. I set canreinvite=yes in sip.conf in both sip peers. I also tested it with 2 mgcp phones and same result, all rtp traffic goes through Asterisk. Is there any way to force traffic to go from one phone to another? snip I don't recall where it is off-hand but, somewhere in the Asterisk documentation, there is an explanation of how Asterisk makes a decision about reinvites. You may want to look at that to see if your environment satisfies all the requirements and how it can be adapted if it does not - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Multi Tenant Asterisk Server ?
On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Multi Tenant Asterisk Server ?
On Sat, 2009-11-14 at 00:30 +0100, Michiel van Baak wrote: On 17:54, Fri 13 Nov 09, John A. Sullivan III wrote: On Fri, 2009-11-13 at 19:55 +, Gavin Spurgeon wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi List, What I hope is a simple question... As the subject states, I would like to know if anyone has setup a Multi Tenant Asterisk Server ? If so, what would I need to do to get to a Multi Tenant setup (preferably an Open Source solution) ? Any suggestions/comments/pointers/URLs ? snip Entirely doable and reasonably well documented in the literature. Pay particular attention to the use of contexts. If I recall correctly, the followme and meetme applications do not support contexts. I believe you also have to be careful with SIP ids even in different contexts (someone correct me on that if I'm wrong as Asterisk is only a small part of my job and so the details are not always fresh in my mind). For those, we rely upon some other globally unique attribute, e.g., in our environment, all tenants have a unique posix uid and username. We use that username for the SIP ID and the uid for the meetme and followme identifiers. Hope this helps - John PS - Ah - multi-tenant parking - it is broken as recently as 1.6.1.7. There is a patch which works perfectly. I do not know if that patch was included in 1.6.1.8. In fact, if someone knows, please respond as we need to do that upgrade for security purposes and are concerned about breaking multi-tenant parking. That patch is not yet in. I'm planning to get it in this weekend. snip Thanks for the update. How will it be available at that point? Will there be an immediate 1.6.1.9 release or will it only be via SVN? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Unable to set TOS to 184?
On Fri, 2009-10-30 at 09:53 +0100, Karsten Wemheuer wrote: Hi Bart, Am Donnerstag, den 29.10.2009, 16:36 -0700 schrieb Bart Fisher: I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 You did not tell us, which version of asterisk You are running. The kernel restricts setting of high ToS-bits for non-root users. To allow a process to run as non-root and be able to set these bits, there is the possibility to use 'capabilities'. This feature was implemtented and fixed in the past (issues 7074 and 14004 at issues.asterisk.org). I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef In case Your iptables is working, I think You can ignore the warning. snip I'm assuming this is working for non-root users in 1.6.1.6. I'm pretty sure the last time I took a packet trace, TOS was being set properly and I am not running as root - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Unable to set TOS to 184?
On Thu, 2009-10-29 at 16:36 -0700, Bart Fisher wrote: I don't understand this message: [2009-10-29 16:31:51] WARNING[28510]: rtp.c:1997 ast_rtp_settos: Unable to set TOS to 184 From what I have read the reason is asterisk can't set TOS if not running in root. Mine is running as asterisk. I found one post that says to run at boot: #!/bin/bash /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 4569 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 1:2 -j DSCP --set-dscp-class ef /sbin/iptables -A OUTPUT -t mangle -p udp -m udp --sport 5060 -j DSCP --set-dscp-class ef Does this make sense? Is this the only method to end ths warning? snip I'm pretty new to Asterisk so take this with a grain of salt. Is it possible you used decimal (184) instead of hex notation (b8) in your sip.conf? We're running 1.6.1.6 and it appears to be working just fine. Here are the pertinent lines from our sip.conf: tos_audio=0xb0 ; b8 (expedited forwarding) confuses the Linux pfifo_fast so b0 works better for us tos_sip=0xb0 The comment is also important in light of the iptables rules you have. As someone else pointed out, you shouldn't need both. I prefer to set them in the application. For example, if I ever change ports for whatever reason, I won't have the problem of forgetting to also change my iptables rules. Now, I may be wrong about this so I wouldn't mind feedback from someone who know better than I do, but I think expedited forwarding (ef = 184 = b8) can shoot you in the foot in Linux. If you don't change the default packet queueing from pfifo-fast, I believe it will not look at the DSCP bits but rather the ToS bits and will place ef packets into band1 (normal priority) rather than band0 (high priority). That's why we use b0 instead and then tell our DSCP enabled switches to place the resultant DSCP values into the highest priority queue. Hope that makes sense. If I'm wrong, please, someone call me out on it. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Answer call from another device
On Mon, 2009-10-26 at 14:58 -0500, Danny Nicholas wrote:
*8 is the default value in features.conf to pick up a ringing line if you
are in that ring group.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Monday, October 26, 2009 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Answer call from another device
Hello!
I remember a while back I saw a way to answer a call from a device
that is not from the one ringing, but I don't remember what how to do
it. Any help would be great!
snip
One can take it even a little further than that depending on the phone.
We set up hints in extensions.conf so the programmable button lights on
our Snom phones indicate if a remote phone is ringing or off hook.
While it is ringing, a user can press the button which will issues the
*8 + extension and pick up the call. Here is an abbreviated example:
; Joe
exten = 613,hint,SIP/joe
; Mary
exten = 614,hint,SIP/mary
; Mike
exten = 616,hint,SIP/mike
; Enable call pickup for hinted stations from any possible source contexts
exten =
_*8XXX,1,Pickup(${EXTEN:2...@a10pub${EXTEN:2...@a10a${EXTEN:2...@a10f)
We're pretty new to Asterisk so there may be a much better way but this
worked for us. Good luck - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] Incorrect voice mail format on transfer
Phew! So it's not just me! That's exactly the problem - not leaving the message but forwarding it (I suppose the correct term rather than transfer). Thanks - John On Thu, 2009-10-22 at 10:29 -0500, Robert Grignon wrote: I did run into some issues with this as well. I ended up setting format=wav and left it at that... It wasn't so much a problem with someone leaving a message rather when someone was forwarding messages. I would have used wav49 but people were having problems getting wav49 to open on their PDA's -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Wednesday, October 21, 2009 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Incorrect voice mail format on transfer I'm sorry - by the lab I meant the end points - it is the same server. I was not aware that IMAP only stored one format. If I change the setting in voicemail.conf, do I still have to worry about the grievous warning message about being sure to delete all messages not using that format? I would think not but it's a dire enough message that I thought I had better ask - John On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote: It should be reproducible in some way, how was asterisk installed on the server its having a problem? If its from source compare the apps/app_voicemail.c from whats in production with whats getting compiled in the lab. when imap is used only one format is stored you could specify just one format: format=wav49 On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email the voicemails in this case (this client is not using the Zimbra email system yet) and they receive an attachment with a name such as msg.wav49_gsm_wav. As strange as it sounds, it almost appears like Asterisk is trying to create a file with an extension of wav49|gsm|wav which is confusing not only the email attachment but also sox which cannot find such a format based upon file extension. Here is what I see in /var/log/asterisk/messages. First, the user doing the transfer: [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed
Re: [asterisk-users] Poor VoIP voice quality in one direction from three providers
On Thu, 2009-10-22 at 16:04 -0700, Robert L Mathews wrote: We currently use asterisk 1.4.x with two Zaptel cards connected to POTS lines. So we make outbound calls from their softphones (using ulaw format), which go over a dedicated DSL line to the asterisk server in our office, which then converts the calls to POTS. This all works fine, assuming there aren't any unusual problems. It sounds as good as POTS on both ends. However, we don't want to maintain the DSL line or deal with the hassles of analog/digital conversion any more. So we want to switch to a reliable VoIP provider and move the asterisk server to one of our colocation data centers. We've tried getting test accounts with three VoIP providers: FlowRoute, CallCentric, and Vitelity. In our tests, outbound calls now go from softphones - asterisk - VoIP provider - outside world. We use ulaw all the way through. But with all three providers, we see a curious thing: The audio quality in the direction from our softphones to the outside world still sounds as good as POTS, but the audio quality in the inbound direction (outside world - VoIP Provider - asterisk - softphone) is noticeably worse. It sounds overcompressed or slightly robotic somehow, with a decrease in dynamic range. It's not lagged or echoey; it just sounds like it's maybe using a crappier codec than ulaw, in that direction only. I'm baffled by this. Both legs of the calls show as Format: 0x4 (ulaw) in sip show channel. Testing the first provider, I just assumed that their analog-digital conversion was inferior to what the Zaptel cards offer (i.e., that they were injecting inferior sound quality into their ulaw connection)... but we're getting exactly the same results with all three providers, which makes me think it's us. Why might this happen? Is there any possible reason other than all three of the VoIP providers are decreasing the audio quality before injecting it into the ulaw stream? I don't know if it is the same issue but we had just the reverse problem and only with softphones. The inbound quality from Vitelity was excellent but the outbound was horrible. After beating on the problem for weeks, tweaking all aspects of both the network (packet prioritization) and kernel (process prioritization), we achieved only marginal improvement. It finally turned out to be the headsets. We had bought mid-range Logitech headsets (actually the most expensive ones from our local retailer). Once we swapped them out for Plantronics Audio 655 headsets, the problems went away - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Incorrect voice mail format on transfer
:03] WARNING[13572] file.c: Unable to open /var/spool/asterisk/voicemail/a10/612/INBOX/msg (format 0x4 (ulaw)): No such file or directory [Oct 21 13:29:03] WARNING[13572] app_voicemail.c: Playback of message /var/spool/asterisk/voicemail/a10/612/INBOX/msg failed I've not been able to reproduce it in our lab but I can see and hear it plainly happening for our client. Has anyone else seen this? Is it a bug or a misconfiguration? voicemail.conf has format=wav49|gsm|wav Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Incorrect voice mail format on transfer
I'm sorry - by the lab I meant the end points - it is the same server. I was not aware that IMAP only stored one format. If I change the setting in voicemail.conf, do I still have to worry about the grievous warning message about being sure to delete all messages not using that format? I would think not but it's a dire enough message that I thought I had better ask - John On Wed, 2009-10-21 at 14:02 -0700, Kyle Kienapfel wrote: It should be reproducible in some way, how was asterisk installed on the server its having a problem? If its from source compare the apps/app_voicemail.c from whats in production with whats getting compiled in the lab. when imap is used only one format is stored you could specify just one format: format=wav49 On Wed, Oct 21, 2009 at 1:19 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I'm running Asterisk 1.6.1.6 on CentOS 5.3 in a multi-tenant environment with IMAP voice mail storage on Zimbra. One of our clients is having a problem when transferring voice mails from one mailbox to another (option 8 in the standard voice application menu) using their Snom 320 and 360 phones. The end results is the final recipient cannot listen to the voicemail. We also email the voicemails in this case (this client is not using the Zimbra email system yet) and they receive an attachment with a name such as msg.wav49_gsm_wav. As strange as it sounds, it almost appears like Asterisk is trying to create a file with an extension of wav49|gsm|wav which is confusing not only the email attachment but also sox which cannot find such a format based upon file extension. Here is what I see in /var/log/asterisk/messages. First, the user doing the transfer: [Oct 21 12:23:17] WARNING[13297] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:27:13] WARNING[13303] app_voicemail.c: IMAP Warning: SECURITY PROBLEM: insecure server advertised AUTH=PLAIN [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:28:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg.WAV: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001.wav49|gsm|wav: No such file or directory [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Sox failed to reencode /var/spool/asterisk/voicemail/a10/613/INBOX/msg0001intro.wav49|gsm|wav: An error occurred during file processing (have you installed support for all sox file formats?) [Oct 21 12:29:44] WARNING[13303] app_voicemail.c: Voicemail attachment will have no volume gain. [Oct 21 12:29:44] WARNING[13303
Re: [asterisk-users] IMAP voicemail using subfolders fails.
On Mon, 2009-10-19 at 17:29 -0400, Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am running 1.6.0.15 and am trying to get IMAP storage working. I have had no trouble doing so, except that I wish to create a subfolder in my account for voicemail, such that I have: #voicemail/ INBOX Old Family Friends Work I can set IMAPFOLDER=#voicemail.INBOX in voicemail.conf and successfully get voicemail to work as expected. Messages appear in INBOX and are deleted if removed from the phone. If, however, I attempt to change folders with option 2, I get the following error: file.c:950 ast_streamfile: Unable to open vm-#voicemail.INBOX (format 0x4 (ulaw)): No such file or directory Clearly, app_voicemail is looking for vm-INBOX and is building the voicemail prompt file name based upon the voicemail folder. I attempted to symlink vm-INBOX.gsm to vm-vm-#voicemail.INBOX.gsm but that didn't help, either. I have tried using the combination of: IMAPPARENTFOLDER=#voicemail IMAPFOLDER=INBOX but in this case the VM system can't find the messages and the voicemail app simply dies. So that isn't the right incantation. Surely, it's possible to do what I'm looking to do, isn't it? So my questions are: How do I configure app_voicemail to use IMAP subfolders? and I have used '/' as the delimter as well as the '.' character. Am I using the wrong one and if so, what is the correct one? snip I can't help you directly but I can share my experience with folders. I intentionally did not set up the folder structure in IMAP as recommended in the documentation. To my pleasant surprise, when the folders were needed (e.g., a user moves a voice mail via the voicemail application to friends, etc.), they were created on the fly, i.e., Asterisk created them within the IMAP folder system. I am using 1.6.1.6 with Zimbra as the backend - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] IMAP voicemail using subfolders fails.
On Mon, 2009-10-19 at 19:08 -0400, Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 John A. Sullivan III wrote: I can't help you directly but I can share my experience with folders. I intentionally did not set up the folder structure in IMAP as recommended in the documentation. To my pleasant surprise, when the folders were needed (e.g., a user moves a voice mail via the voicemail application to friends, etc.), they were created on the fly, i.e., Asterisk created them within the IMAP folder system. I am using 1.6.1.6 with Zimbra as the backend - John Thanks, John. I didn't see that in the docs. I am going to do what you suggested and just let Asterisk put things in the root directory. Did you perhaps use the INBOX or are you using a custom folder? snip I'm using the INBOX - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] MWI for multiple voice mail boxes
On Sun, 2009-10-18 at 19:14 -0500, Tilghman Lesher wrote: On Thursday 15 October 2009 20:13:55 John A. Sullivan III wrote: On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John No, comma is the right delimiter, unless you're using ODBC storage for voicemail, in which case, I'm terribly sorry, but multiple mailboxes are not supported in that line. This has been corrected in SVN for all 1.6 branches. I'm not using ODBC but I am using IMAP. Could that be the problem? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk Monitoring
On Sun, 2009-10-18 at 01:30 +0100, Dan Journo wrote: Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. snip It depends on how deep and far you want to go. As already mentioned, Nagios will do a nice job of letting you know SIP is responding and is an all around excellent product. Despite our familiarity with Nagios, we've been taking the plunge into OpenNMS. It is much, much, much more difficult to set up for Asterisk monitoring and is an enormous product. On the other hand, it gives you access into the entire Asterisk SNMP MIB, does an outstanding job of collecting and presenting statistics for all channels (not just SIP - Nagios could theoretically expose the entire MIB but I don't know that anyone has written such a plugin and doing it via the SNMP plugin alone would be difficult at best). We were very keen to track much more data than Nagios gave us in order to tweak, troubleshoot, and diagnose our services. Hence the investment in learning OpenNMS. Once you get past the huge learning curve of OpenNMS, it is a remarkably powerful product. There was a flurry of queries regarding the specifics of setting up Asterisk monitoring on the OpenNMS mailing list just a couple of days ago and another string on monitoring SIP a couple of weeks ago. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
No, probably my ignorance but why would I do that? I set up all the users, extensions, and mailboxes manually by editing the config files in order to have more control than the user.conf gives me (if I understand the user.conf file properly - I've never used it based upon reading the documentation). Thanks - John On Fri, 2009-10-16 at 12:07 -0500, Danny Nicholas wrote: I assume you have a 610 entry in users.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Friday, October 16, 2009 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes Alas, it does not work for me on 1.6.1.6. That was my original configuration based upon the documentation. It was slightly different than you have because I specified the context. tkeeley is in context a10f but the mailboxes are in context a10. Thus, I had: [tkeeley] mailbox=...@a10, 6...@a10 It then complains that it cannot find mailox 610 in context a10. However, it is there and it does receive voice mail. Thanks - John On Fri, 2009-10-16 at 10:14 -0500, Danny Nicholas wrote: Let's stick a fork in this one - Here's the link I used http://www.voip-info.org/wiki/view/Asterisk+sip+mailbox if we make tkeely's sip.conf look like this [tkeeley] Type=peer Context=a10 Mailbox=612, 610 He? Should be good to go. This worked on 1.4.26.1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 8:14 PM To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Where to find IMAP storage doc ?
On Thu, 2009-10-15 at 10:52 -0400, Matthew Harrell wrote: Where can I find doc related to IMAP storage. Usually, config options can be found either in voicemail.conf or voip-info.org but almost none relates to IMAP configuration. At the moment, I'm looking for data related to imapflags possible values. More or less everything I know I found on this old email http://www.mail-archive.com/courier-us...@lists.sourceforge.net/msg27564.html It worked fine at the time although, I have to admit, I don't know if I've even tried to use it in the last 6 months We're also working fine with it but I also do not know what the available imapflags are and what they mean. I have seen notls and novalidatecert. Out of curiosity, I spent the last 20 minutes googling for information on c-client imapflags and didn't find any definitions or even a simple list, either. There is a list of flags in the c-client man page but they seem to be a different set of flags. Let me know what you find as I would like to know what functionality and options they give us. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
Ah, interesting. I wasn't aware that it only used the first value. What's the purpose of the secondary values then? If I understand you correctly, you are saying I should have one entry for tkeeley with two entries for mailbox=? Thanks - John On Thu, 2009-10-15 at 12:54 -0500, Danny Nicholas wrote: Just a thought... If the SNOM has multiple lines, tying one to 612 and the other to 610 should make the MWI active for both lines. Asterisk AFAIK only actives the first entry in the list, so you would need two entries for tkeeley with mailbox=612 in the first instance and mailbox=610 in the second. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A. Sullivan III Sent: Thursday, October 15, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI for multiple voice mail boxes On Thu, 2009-10-15 at 19:24 +0200, Olivier wrote: 2009/10/15 John A. Sullivan III jsulli...@opensourcedevel.com Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 From memory, I could successfully make this happen (1 MWI for several mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? Actually, I've not tried reversing them. We are in production so I'll need to wait until tonight to test. Thanks - John However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MWI for multiple voice mail boxes
On Thu, 2009-10-15 at 15:29 -0700, Jared Smith wrote: On Wed, 2009-10-14 at 22:41 -0400, John A. Sullivan III wrote: Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 I think you've got the syntax wrong here... try mailbox=...@a106...@a10 instead. Contrary to what others on this thread might lead you to believe, this should actually work. :-) snip O - it really didn't like that: mailbox=...@a106...@a10 app_voicemail.c:1630 messagecount: Couldn't find mailbox 612 in context a106...@a10 It looks like it's interpreting everything after the @ as context. I'm running 1.6.1.6. Thanks anyway - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Callpickup works for outside calls but not inside calls
On Wed, 2009-10-14 at 22:56 -0400, John A. Sullivan III wrote:
Hello, all. I've got a problem where we set up call pickup for a
customer. If the Bob's extension rings and Bob is in Jim's office, Bob
can press the button on his Snom 320 that says Bob and pick up his
line. It works great for calls coming in from the outside but does not
work for internal calls. Internal calls generate a
app_directed_pickup.c:204 pickup_exec: No target channel found for 617
error.
I see an old bug about this where the contexts were not consistent but
ours appear to be consistent. Here are examples of pertinent parts of
the dialplan:
[a10base]
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
; Terry Keeley
; We put these in a10base rather than a10 or a10pub
;so that the spare stations can access them but public cannot
exten = 612,hint,SIP/tkeeley
; Joe Intrabartola
exten = 613,hint,SIP/jintrabartola
; Maryann Lapolla
exten = 614,hint,SIP/mlapolla
; Michael Intrabartola
exten = 616,hint,SIP/mintrabartola
; Vinny De Marco
exten = 617,hint,SIP/vdemarco
; Reception - the Reception desk may ring when someone dials zero
exten = 621,hint,SIP/reception-a10
; Steve McClain
exten = 624,hint,SIP/smcclain
; Amityville Intercom
;exten = 686,1,Dial(SIP/avilleextdoor-a10,60)
;exten = 686,n,Hangup()
exten = _*8XXX,1,Pickup(${EXTEN:2...@a10pub) ; Enable call pickup for hinted
stations
exten = 7998,1,VoiceMailMain(${CALLERID(num)}...@a10) ; Direct mail retrieval
exten = 7998,n,Hangup()
include = a10pub
include = a10utils
include = a10conf
include = a10parking
[a10in] ; direct inbound SIP dialing
exten = conference,1,Goto(a10pub,6000,1)
exten = joe,1,Goto(a10pub,613,1)
exten = maryann,1,Goto(a10pub,614,1)
exten = michael,1,Goto(a10pub,616,1)
exten = terry,1,Goto(a10pub,612,1)
exten = tommyvan,1,Goto(a10pub,615,1)
exten = vinny,1,Goto(a10pub,617,1)
exten = ebc,1,Goto(a10pub,9,ringall)
exten = vmail,1,Goto(a10pub,7999,1)
[a10pub]
; Public access - BE SURE there is no outbound access from here, e.g.,
; Background() functions will jump to any valid extension entered
; whether or not it is listed in the menu
; Terry Keeley
exten = 612,1,Set(__VM=612) ; VoiceMail ID
exten = 612,n,Gosub(a10ringtones,internal,1)
exten = 612,n,Macro(common,SIP/tkeeley,1,a10)
; 1 for VM, a10 VM context, no followme, ring for default seconds
exten = 8612,1,VoiceMail(6...@a10,u)
exten = 7612,1,VoiceMailMain(6...@a10)
exten = 7612,n,Hangup()
; Joe Intrabartola
exten = 613,1,Set(__VM=613)
exten = 613,n,Gosub(a10ringtones,internal,1)
exten = 613,n,Macro(common,SIP/jintrabartola,1,a10)
exten = 8613,1,VoiceMail(6...@a10,u)
exten = 7613,1,VoiceMailMain(6...@a10)
; Vinny De Marco
exten = 617,1,Set(__VM=617)
exten = 617,n,Gosub(a10ringtones,internal,1)
exten = 617,n,Macro(common,SIP/vdemarco,1,a10)
exten = 8617,1,VoiceMail(6...@a10,u)
exten = 7617,1,VoiceMailMain(6...@a10)
; Floral Park Spare
exten = 618,1,Gosub(a10ringtones,internal,1)
exten = 618,n,Dial(SIP/sparef1-a10,120,o) ; Ring the phone for up to 2
minutes
exten = 618,n,Hangup()
If I make a SIP call across the Internet to Vinny, for example, we issue
a goto to Vinny's internal extension. Terry can press the call pickup
and it all works. The same if I dial in from the PSTN. Here is the
call sequence:
-- Executing [vi...@a10in:1] Goto(SIP/jasiii-ad0e1048, a10pub,617,1)
in new stack
-- Goto (a10pub,617,1)
-- Executing [...@a10pub:1] Set(SIP/jasiii-ad0e1048, __VM=617) in new
stack
-- Executing [...@a10pub:2] Gosub(SIP/jasiii-ad0e1048,
a10ringtones,internal,1) in new stack
-- Executing [inter...@a10ringtones:1]
SIPAddHeader(SIP/jasiii-ad0e1048, Alert-Info:
http://www.notused.com\;info=alert-internal\;x-line-id=0) in new stack
-- Executing [inter...@a10ringtones:2] Return(SIP/jasiii-ad0e1048, )
in new stack
-- Executing [...@a10pub:3] Macro(SIP/jasiii-ad0e1048,
common,SIP/vdemarco,1,a10) in new stack
-- Executing [...@macro-common:1] Set(SIP/jasiii-ad0e1048, TM=24) in
new stack
-- Executing [...@macro-common:2] Dial(SIP/jasiii-ad0e1048,
SIP/vdemarco,24,o) in new stack
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
-- Called vdemarco
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
-- SIP/vdemarco-d4012df8 is ringing
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
== Extension Changed 612[a10base] new state InUse for Notify User
jintrabartola
== Extension Changed 612[a10base] new state InUse for Notify User
reception-a10
-- Executing [*8...@a10f:1] Pickup(SIP/tkeeley-acc9aaf8, 6...@a10pub)
in new stack
== Extension
Re: [asterisk-users] Door Phones
On Wed, 2009-10-14 at 14:05 -0700, Jonathan Thurman wrote: On Wed, Oct 14, 2009 at 1:52 PM, Dan Journo d...@keshercommunications.com wrote: Hi, Can anyone recommend a cheap SIP doorphone? Please only respond if you’ve had personal experience of a doorphone. I searched around for a while and couldn't find a hardened SIP external phone. We ended up using an ATA and a regular outside door phone from Ceeco (http://www.ceeco.net/ Model WPP-531-F). For a analog phone in a metal box, they aren't exactly cheap. You could say that an Analog phone would be more secure if someone ripped it off the wall, they wouldn't have network access. Then you just lock down what numbers can be called on your PBX. snip We've just installed a CyberData VoIP intercom and are quite happy with it so far: http://www.cyberdata.net/products/voip/digitalanalog/intercom/index.html -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] MWI for multiple voice mail boxes
Hello, all. I have a user who needs to monitor their voice mail box and the general delivery voice mail box. I defined them in sip.conf as follows: [tkeeley](a10f) mailbox=...@a10, 6...@a10 However, the MWI does not indicate voice mails for 610 and I keep seeing this error message: ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox 610 in context a10 However, mailbox 610 is clearly defined in voicemail.conf: [a10] 610 = xxx,General Mailbox,m...@mycompany.com,,imapuser=m...@mycompany.com 612 = yyy,Terry Keeley,morem...@mycompany.com,,imapuser=morem...@mycompany.com The end device is a Snom 360. We are running Asterisk 1.6.1.6. Why are we receiving this error when the mailbox is clearly defined? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Callpickup works for outside calls but not inside calls
User sparea1-a10 == Extension Changed 612[a10base] new state InUse for Notify User clerical-a10 == Extension Changed 612[a10base] new state InUse for Notify User confroom-a10 == Extension Changed 612[a10base] new state InUse for Notify User smcclain == Extension Changed 612[a10base] new state InUse for Notify User vdemarco -- Executing [*8...@a10f:2] Playback(SIP/tkeeley-acc9aaf8, im-sorry) in new stack -- SIP/tkeeley-acc9aaf8 Playing 'im-sorry.ulaw' (language 'en') NOTE PICKUP APPEARS TO FAIL (I'm sorry message) BUT THEN SUCCEEDS -- SIP/tkeeley-acc9aaf8 answered SIP/jasiii-ad0e1048 == Extension Changed 617[a10base] new state Idle for Notify User jintrabartola == Extension Changed 617[a10base] new state Idle for Notify User confroom-a10 -- Packet2Packet bridging SIP/jasiii-ad0e1048 and SIP/tkeeley-acc9aaf8 == Spawn extension (a10f, *8617, 2) exited non-zero on 'SIP/vdemarco-d4012df8ZOMBIE' == Extension Changed 617[a10base] new state Idle for Notify User sparea1-a10 -- Executing [...@a10f:1] Hangup(SIP/vdemarco-d4012df8ZOMBIE, ) in new stack If we dial Vinny's extension from an internal phone, say sparef1-a10, and Terry tries to pick it up, it fails but the call sequence looks identical except the pickup never bridges the call and generates an error. Here is the nearly identical sequence: -- Executing [...@a10f:1] Set(SIP/sparef1-a10-ad0b12f8, __VM=617) in new stack -- Executing [...@a10f:2] Gosub(SIP/sparef1-a10-ad0b12f8, a10ringtones,internal,1) in new stack -- Executing [inter...@a10ringtones:1] SIPAddHeader(SIP/sparef1-a10-ad0b12f8, Alert-Info: http://www.notused.com\;info=alert-internal\;x-line-id=0) in new stack -- Executing [inter...@a10ringtones:2] Return(SIP/sparef1-a10-ad0b12f8, ) in new stack -- Executing [...@a10f:3] Macro(SIP/sparef1-a10-ad0b12f8, common,SIP/vdemarco,1,a10) in new stack -- Executing [...@macro-common:1] Set(SIP/sparef1-a10-ad0b12f8, TM=24) in new stack -- Executing [...@macro-common:2] Dial(SIP/sparef1-a10-ad0b12f8, SIP/vdemarco,24,o) in new stack == Using SIP RTP TOS bits 176 == Using SIP RTP CoS mark 5 -- Called vdemarco -- SIP/vdemarco-1c41fc38 is ringing -- SIP/vdemarco-1c41fc38 is ringing -- SIP/vdemarco-1c41fc38 is ringing -- SIP/vdemarco-1c41fc38 is ringing -- SIP/vdemarco-1c41fc38 is ringing == Using SIP RTP TOS bits 176 == Using SIP RTP CoS mark 5 == Extension Changed 612[a10base] new state InUse for Notify User jintrabartola -- Executing [*8...@a10f:1] Pickup(SIP/tkeeley-acc4df68, 6...@a10pub) in new stack == Extension Changed 612[a10base] new state InUse for Notify User reception-a10 [Oct 14 22:14:41] NOTICE[2778]: app_directed_pickup.c:204 pickup_exec: No target channel found for 617. -- Executing [*8...@a10f:2] Playback(SIP/tkeeley-acc4df68, im-sorry) in new stack == Extension Changed 612[a10base] new state InUse for Notify User mintrabartola == Extension Changed 612[a10base] new state InUse for Notify User mlapolla == Extension Changed 612[a10base] new state InUse for Notify User sparea1-a10 == Extension Changed 612[a10base] new state InUse for Notify User clerical-a10 == Extension Changed 612[a10base] new state InUse for Notify User confroom-a10 == Extension Changed 612[a10base] new state InUse for Notify User smcclain == Extension Changed 612[a10base] new state InUse for Notify User vdemarco -- SIP/tkeeley-acc4df68 Playing 'im-sorry.ulaw' (language 'en') -- Executing [*8...@a10f:3] Wait(SIP/tkeeley-acc4df68, 0.0.5) in new stack -- Executing [*8...@a10f:4] Playback(SIP/tkeeley-acc4df68, you-dialed-wrong-number) in new stack -- SIP/tkeeley-acc4df68 Playing 'you-dialed-wrong-number.ulaw' (language 'en') -- Executing [*8...@a10f:5] Wait(SIP/tkeeley-acc4df68, 0.4) in new stack -- Executing [*8...@a10f:6] Playback(SIP/tkeeley-acc4df68, vm-goodbye) in new stack -- SIP/tkeeley-acc4df68 Playing 'vm-goodbye.ulaw' (language 'en') -- Executing [*8...@a10f:7] Hangup(SIP/tkeeley-acc4df68, ) in new stack == Spawn extension (a10f, *8617, 7) exited non-zero on 'SIP/tkeeley-acc4df68' -- Executing [...@a10f:1] Hangup(SIP/tkeeley-acc4df68, ) in new stack == Spawn extension (a10f, h, 1) exited non-zero on 'SIP/tkeeley-acc4df68' They look identical to me! Except one works and one doesn't. What did I do wrong? How do I configure this so call pickup works for both external and internal calls? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api
Re: [asterisk-users] VoiceMail and IMAP
On Fri, 2009-10-09 at 15:07 +0100, --[ UxBoD ]-- wrote: Hi, I have followed the article on how to install Asterisk with VM in IMAP but for some reason it still continues to send it as a email. I have the following in voicemail.conf :- imapserver= imapfolder=voicemail imapport=143 expungeonhangup=yes imapflags=notls authuser=x authpassword=x and I have added imapuser and imappassword to the configured users. Could it be because I still have their email address specified and that is overriding it ??? snip Hi, Phil. I believe that is the case exactly. In fact, we have a hybrid case. Some of our clients are using our mail system in which case their messages appear natively in their Zimbra account. Others are not so we simply email them in the traditional way. Take care - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Best QoS for Linux
On Thu, 2009-10-08 at 16:07 -0400, Michelle Dupuis wrote: More specificallyI'm looking for a Linux package to allow shaping, QoS, prioritization by port, etc. snip Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? _snip I would imagine that tc, iproute2, and iptables are your friends. In our case, we try to keep things as simple as possible in a fairly complex environment. Thus, whenever we can, we try to set our DSCP/ToS bits in a way that will be handled properly by the default Linux queueing mechanism. I'm afraid I'm up to my eyeballs in a project right now but I have posted some of our work in earlier posts on this mailing list. In the case of Asterisk, we use b0 instead of b8 (expedited forwarding) for RTP traffic because it works better with the default pfifo_fast packet scheduler. We've also ensured the packet handling is consistent from end to end as much as possible. Even though we are using the Internet as a transport medium, we're very happy so far with the quality of the calls. See the previous posts for more details. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Help setting up IMAP_STORAGE on CentOS 5
On Thu, 2009-10-08 at 15:45 -0500, Jason Parker wrote: Noah I. Engelberth wrote: I’ve been spending the day trying to get IMAP_STORAGE on my test box, to evaluate for production, but I’m having no luck getting uw-imap to build. I’ve tried installing it from an upstream package, but Asterisk still isn’t finding it to compile –with-imap. My google searches have turned up very little for documentation on dependencies, gotchas, etc for either item, so I’m hoping someone here can help me get IMAP set up for my Asterisk box. You should be able to just `yum install libc-client-devel` on CentOS. snip That should indeed solve the dependency problem but it doesn't always work when it comes time to connect to the IMAP server. We spun our wheels for hours and hours trying to figure out why. As it turned out in our case, the version of libc-client included in CentOS was too old to work with our Zimbra 5.0.16 installation. We had to install cclient from source. It may or may not be a problem for Noah but it certainly bit us! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] app_hackblock to prevent SIP/IAX reg trolling
Is that what that does? I assumed that was like a protocol retry. In other words, if the registrar does not reply to the registry when it submits its credentials, it will resubmit them registerattempts number of times. I did not think that prevented a registree from submitting 10,000 new sets of credentials. But that was only my guess - John On Fri, 2009-10-02 at 14:58 -0500, Danny Nicholas wrote: Sipregisterattempts would seem to be the simplest way to do this. It is 0 by default, changing it to 5 would stop the hacker after 5 tries. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Friday, October 02, 2009 2:24 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling Good post. One of the recommendations is to limit the number of calls per sip entity. Is there an easy way to do that in sip.conf? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Friday, October 02, 2009 3:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] app_hackblock to prevent SIP/IAX reg trolling Couple of old posts: http://lists.digium.com/pipermail/asterisk-users/2007-April/186195.html http://lists.digium.com/pipermail/asterisk-users/2009-March/229479.html http://lists.digium.com/pipermail/asterisk-users/2007-April/186456.html On Fri, Oct 2, 2009 at 2:42 PM, Michelle Dupuis supp...@ocg.ca wrote: Has anyone written an app that monitors SIP/IAX registration attempts? A couple of clients are being flooded with SIP registrations (but the source IP changes every few hours so IPtables won't do).. I would think that any attempt to reg 5 times with a bad password should cause a 5 minute timeout until reg is considered again. Has anyone written such an app? The name app_hackblock is my contribution to the project :) MD ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] What are the reasons for VoIP echo?
I'm quite new to all this but I was under the impression that most electrically induced echo was at the physical interface to the PSTN. If one is using SIP trunking, I would think this would point to a carrier issue. We also hit an interesting problem with echo today but I don't think this is the issue Myles is having. We installed fairly high end phones with full duplex speakerphones. Callers are having a bad problem with echo when the users use the speakerphone. Because it is full duplex rather than half, if the speakerphone volume and speakerphone mike volume are turned up, the callers are indeed hearing themselves by virtue of the higher quality full duplex! On Thu, 2009-10-01 at 19:36 -0500, Martin wrote: if a user calling you hears echo of himself then it's the fault of your sip device/sip phone. The manufacturer must be using a cheap or an open source echo canceller ... try getting a different sip device made by some 'normal' company like polycom or linksys/cisco Martin On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham my...@techsol.org wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
Re: [asterisk-users] What are the reasons for VoIP echo?
Indeed there are! - John On Thu, 2009-10-01 at 20:18 -0500, Martin wrote: Are you saying there are half duplex phones out there with half duplex speakerphones ? All analog phones are full duplex ... Anyways the echo can be created by the analog phone even when it's connected to the sip ata or even the sip phone ... then you usually have acoustic echo which goes from speaker to microphone of the handset ... that should be cancelled by the sip phone/device... or someone out there will hear echo Martin On Thu, Oct 1, 2009 at 7:57 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: I'm quite new to all this but I was under the impression that most electrically induced echo was at the physical interface to the PSTN. If one is using SIP trunking, I would think this would point to a carrier issue. We also hit an interesting problem with echo today but I don't think this is the issue Myles is having. We installed fairly high end phones with full duplex speakerphones. Callers are having a bad problem with echo when the users use the speakerphone. Because it is full duplex rather than half, if the speakerphone volume and speakerphone mike volume are turned up, the callers are indeed hearing themselves by virtue of the higher quality full duplex! On Thu, 2009-10-01 at 19:36 -0500, Martin wrote: if a user calling you hears echo of himself then it's the fault of your sip device/sip phone. The manufacturer must be using a cheap or an open source echo canceller ... try getting a different sip device made by some 'normal' company like polycom or linksys/cisco Martin On Thu, Oct 1, 2009 at 9:10 AM, Myles Wakeham my...@techsol.org wrote: I have an Asterisk 1.4.2 system that has been installed for about 3 months now in our home. We converted all of our phones to SIP phones, and use two different trunk providers (BroadVoice for incoming FlowRoute for outgoing). Most of the time its working flawlessly. But about 1/3rd of the calls that come into us complain of an echo and what is best described as latency issues. Its not consistent though. I was on the phone with an insurance company yesterday for about 1 hour and the call was perfect (I originated the call which used Flowroute for the SIP provider). What seems to be a pattern here is cell phones. When we receive a call from a cell phone, or from certain people on certain phone systems, they consistently complain of echo in the call. Its far less regular when we originate the call, which suggested to me that the problem might be with Broadvoice. But I'm now hearing that us calling back the party doesn't always solve the problem either. We upgraded our Internet feed (we're on a cable Internet through our cable company, with 12mb/s down, 1.5mb/s up) and that seems to have helped but not solved this problem. From what I can see, its some form of latency issue. We use IPCop as a firewall for our Internet access, but have turned off any IDS on it so that its running fast. I can play online computer games through the network with no issues at all, so I don't think its slowing down the traffic and if it was I'd expect this problem to be occurring consistently on all calls. Are there any tweaks that I can do with Asterisk to increase the network performance to reduce these issues? Have others who have experienced this been able to identify the issues to external VoIP SIP providers only, or does our system have something to do with all of this? At the time of the calls coming in, IPCop is telling me that we don't have more than 100K/s of bandwidth in use, and according to the network bandwidth graphs there, even with 2 people on the phone at the same time, the bandwidth never seems to exceed 300K/s, so I think we have plenty of headroom for this. I checked with our cable provider for issues with modem latency, and they couldn't detect anything. Again, I'm not experiencing any lag issues with computer games, particularly those that are heavy in interactivity, so I don't think that is the reason. Any suggestions as to what could be tweaked would be greatly appreciated. Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009
Re: [asterisk-users] Music On Hold
I'm afraid I can't be much help as I am both a newbie and it works just fine for me on 1.6.1.6. Of course, mine was a fresh installation. Is there anything in the logs to give you a clue? You see the wav files but do you see the files encoded for the codecs you are using? I think Asterisk will transcode on the fly but I'm not sure. Sorry - John On Wed, 2009-09-30 at 11:52 +0300, Cyprus VoIP wrote: Hello, We posted the question below yesterday, but got no answer from the community. When we checked the same behavior with Asterisk 1.2, we got the Started music on hold, class... message on the console, but in 1.6, we get absolutely nothing. I tried to unload and reload the moh module and everything seems normal, but Asterisk still doesn't respond in the console to the HOLD action, represented by the INVITE message. the call itself is being placed on hold and can be retrieved, but the audio file is not played and the held party hears only a silence. If anyone knows how to debug/fix it, your help would be HIGHLY appreciated. We're really stuck. Thank you all in advance. Original Message Subject: Music On Hold From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, 29 September, 2009 14:31:28 Hello, We need help in debugging Music On Hold on our Asterisk 1.6.1.6 From the SIP debug, I see that an extension sends an INVITE of the call to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but I don't see in the console any reference to the call being placed on hold. When I typed moh show files, I see the wav files of the /var/lib/asterisk/moh folder. How can I debug this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Music On Hold
On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote: snip You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. snip Hmm . . only one wav file. We had several. As I recall now, we actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold music. 1.6.1.6 has the new hold music. But I believe there are several files. Is that wav file valid, i.e., if you copy it to a system with a sound card and play it, does it play? Could it have been corrupted in copying or have incorrect permissions? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] LDAP integration
On Tue, 2009-09-29 at 11:01 -0300, Rafael Seste wrote: Hi all, I looked on the Internet but I didn't find any good how-to. I would like to integrate a ldap server ( with all users data) with asterisk to authenticate SIP users. With this solution I will only need to add a user on ldap, it will not be necessary to add any special configuration on sip.conf Is that possible???If so, How can I configure this setup??? Thanks in advance I considered doing this using LDAP as a real-time database. I decided not to for two reasons which I'll share below. However, I am very new to Asterisk so I would be very curious to know from more experienced folks if my assumptions were false. First, there were some good how-tos about using LDAP as a real-time database but, if I recall, the schema is extended in such a way that the regular user password is not the password used by Asterisk. Second, I believe we saw a way we could map the Asterisk password to the regular user password (it's been a while so I'm not sure about that) but were concerned about the problems of entering secure passwords from a phone keypad. We enforce fairly secure passwords - at least nine characters with some variety of characters and encourage much longer passwords. Having to enter lots of characters in both cases as well as symbols seemed difficult from a phone keypad. Thus, we decided (reluctantly) to use separate simple passwords for phone access instead of the very secure passwords we use to data access. Hope this helps and looking forward to more informed comments than mine! - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Secure passwords, was LDAP integration
On Tue, 2009-09-29 at 11:23 -0500, Tilghman Lesher wrote: On Tuesday 29 September 2009 10:30:37 John A. Sullivan III wrote: Second, I believe we saw a way we could map the Asterisk password to the regular user password (it's been a while so I'm not sure about that) but were concerned about the problems of entering secure passwords from a phone keypad. We enforce fairly secure passwords - at least nine characters with some variety of characters and encourage much longer passwords. Having to enter lots of characters in both cases as well as symbols seemed difficult from a phone keypad. Thus, we decided (reluctantly) to use separate simple passwords for phone access instead of the very secure passwords we use to data access. I would hope that you're at least restricting your peers to be limited to a set of IPs distinctive to your phones. Otherwise, this is a recipe for disaster, especially if a) your registration server is accessible externally, and b) your phones are permitted to make toll calls, especially international numbers. Most good IP phones permit a method of configuration which does not require typing a password into a keypad. You should probably learn to use that method or switch to a phone with that ability, then use secure passwords. Phones are just as important as data and should be supplied with complex passwords. Thanks for the feedback. Indeed, we do restrict the SIP domains and do not allow registration from outside the internal network and we do use passwords - just not as sophisticated. Perhaps I am being overly conscious of client simplicity. I was thinking of the case where internal users might temporarily move to another phone. Rather than pulling up the web interface to the phone, we wanted them to be able to register through the phone keypad. I suppose they would need to enter their IDs anyway and those are alpha-numeric. Thus, the entering passwords would be similar to entering the IDs. On the other hand, we do tend to use the same registration password for voicemail and meetme and those are regularly entered from the key pad. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] New thread - SIP over VPN
On Sat, 2009-09-26 at 19:32 +, Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, Alan Lord (News) wrote: Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al I have a client with Sonicwall VPNs. Asterisk is at head office on internal LAN, six external locations all have Linksys 2102 ATAs and Polycom IP501 phones registering and placing calls through the tunnels. It seems to work fine, but there is plenty of bandwidth between the offices, and they use G729. I think wrapping up the UDP stream into a TCP based tunnel might cause havoc if there is any packet loss or delay. snip We are using SIP over both IPSec and SSL VPNs very successfully with access controls in the tunnel ingress via the ISCS network security management project (http://iscs.sourceforge.net). There are a couple of issues. I'm not sure what you mean by a TCP tunnel unless you are referring to something like using OpenVPN over TCP rather than the default UDP. IPSec tunnels (which we use for LAN-to-LAN connections) are an IP level protocol and not TCP. OpenVPN (which we use for remote access) defaults to UDP port 1194 but can use any UDP or TCP socket. There has been some discussion that using it over TCP for VoIP can produce better results because the packets are less likely to be delivered out of order although perhaps with greater latency. All VPN processes will introduce additional latency. We have not found that to be a problem but several rounds of encryption / decryption over long distance connections in complex environments might introduce enough latency to be problematic. We have not found that yet. Depending on your VPN protocol implementation, there may or may not be an option to pass the ToS bits from the original packet into the IP header of the VPN packet. This is very important. Even though the Internet will not honor the ToS bits, you will want the gateways on both ends to do so, especially the one placing the packets onto your last mile. Since the VPN gateways cannot look inside the packet until it is decrypted, they have no way of distinguishing a large FTP packet from an RTP packet. Passing the ToS bits through may help. However, be careful. Most VoIP implementations seem to be setting DSCP bits instead of explicitly the ToS bits. DSCP uses the ToS bits but in a way different from the way ToS is set up to interpret them. If I remember correctly, setting DSCP to Expedited Forwarding sets the bits which coincide with ToS in such a way that Linux based gateways will place the packets into the band 1 which is the default processing band and not band 0 which is the high priority band. For example, on Asterisk, we did not set our RTP QoS to b8 but rather to b0 (if I recall correctly). We have one case using OpenVPN where the sound quality is occasionally problematic. In our case it's a little easy. The remote desktops are based upon our soon to be released SimplicITy model (http://www.ssiservices.biz) and accessed via NX or X2Go technology. Usually, the only traffic passing through the OpenVPN tunnel is the VoIP traffic. We have thus changed the gateway itself to treat all UDP packets on port 1194 as high priority. We'll see if that makes the problem go away. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] New thread - SIP over VPN
On Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, John A. Sullivan III wrote: snip We are using SIP over both IPSec and SSL VPNs very successfully with access controls in the tunnel ingress via the ISCS network security management project (http://iscs.sourceforge.net). There are a couple of issues. I'm not sure what you mean by a TCP tunnel unless you are referring to something like using OpenVPN over TCP rather than the default UDP. Isn't an SSL based tunnel all TCP? Not in the case of OpenVPN. I'm not sure about the commercial offerings. That could very well be the case as I believe most of them developed out of the web proxy model. I was probably trapped by my own context! Thanks - John snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Choppy sound, SIP calls within LAN
On Fri, 2009-09-25 at 13:01 +0300, andreil1 wrote: Hi! I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE repository). As a clients I use XLite on Mac, all on the same LAN. Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM and plenty of disk space on LEVEL 5 RAID. Calls to another SIP server (also asterisk) hosted by another company are 100% OK, so it is clearly problem with my server setup. Background music (before pickup) runs fine, but transmitted voice sound is very choppy, no matter of which codec I use. I have searched over net, and implemented one by one every reasonable receipt found, including. highpriority = yes internal_timing = yes transmit_silence = no nat = yes localnet=192.168.0.0/255.255.0.0 externip = xx.xx.xx.xx dtmfmode=rfc2833 Downgrading asterisk did not solved problem, too. Anyone please help if possible.. Many thanks in advance for any suggestion(s). snip My first guess would be a network problem. Is there something different in the network path between the users and the hosted Asterisk server versus the users and the internal Asterisk server? Have you implement some form of CoS / QoS internally (one should)? If you run a continuous ping from a user to the internal Asterisk server, is there any packet loss or congestion (indicated by widely varying response times)? Just a few thoughts - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] multiple contexts for multiple locations
On Fri, 2009-09-25 at 16:58 -0500, das sandesh wrote: Hi All, I have a senario where we have multiple locations and all have the ability to call using 1NX pattern, so we have created multiple contexts so the outbound goes fine, but while transfer occurs (after picking the inbound call and transfer), it uses the first 1Nx priority patterned context, like if the 3rd location is making a transfer, but 1st location have the priority since it is declared first..so i am not able to adjust proper priorities based on the context..Is there a way to search based on the extension's context.since the extensions have the contexts based on the locations... snip I'm not sure I fully understand the problem. If it is similar to ours we we needed to match outbound patterns and general sip patterns and have multiple locations and contexts, we enforced the order of extension processing by using include statements - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Parking - How to transfer the other party toagiven slot
Won't that hangup the call after 60 seconds? - John On Wed, 2009-09-23 at 15:22 -0500, Danny Nicholas wrote: Here’s a snippet from a reply from Jared Smith (Digium, Huntsville AL) - untested exten = 11234,1,Set(TIMEOUT(absolute)=60) exten = 11234,n,MeetMe(11234,d1M) This should create a dynamic room 11234 and send the caller to it for 60 seconds. __ From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, September 23, 2009 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Parking - How to transfer the other party toagiven slot 2009/9/23 Danny Nicholas da...@debsinc.com This stands to be corrected, but for your purpose, a dynamic conference is preferable to a parking lot. The Park application is designed to sequentially use/reuse a series of “lots”. By transferring the caller to conference 11234, you would be able to have the agent pick up the call by going to conference 11234. Yes, I think I like this idea ... How do you transfer the remote party to conference 11234 ? (Please, apologize if this question seems stupid but I'm really a newbie on this topic). Is it easy to mimic parking lot timeout feature (to be certain a caller is not left alone in a dynamic conference) ? __ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, September 23, 2009 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Parking - How to transfer the other party to agiven slot Hi, I'm having trouble to figure out how I could implement this feature : When on call with a contact, local operator would dial a sequence which would park the remote party to a specific parking slot, among the hundred of existing slots. (to each extension, a single specific parking slot is attached and there are too many extensions to dedicate BLF or short DTMF sequence to each) . Example: Operator receives a call from 0123456789. Call He talks to remote party and then decides the call is for extension 1234. As extension 1234 is busy at the moment, Operator forwards the incoming call to slot 11234, typing *911234, for instance. The person using extension 1234 would see that slot 11234 is busy and would try to shorten ongoing call. Should I use features.conf's dynamic features for that (to allow a specific DTMF sequence while on call) ? Then how can I let Operator type digits after *91 prefix ? Should I use Incomplete() application ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Call forwarding, callerID, and e911
We were able to solve the below problem. I'll post it in case someone
encounters the same issue. No need to respond or even read unless you
see a better way. Thanks - John
We have manually set callerID on our outbound lines to reflect the
appropriate DID both for e911 and to be polite to folks we call, e.g.:
exten = _1NXXNXX,1,Set(CALLERID(num)=5197546340)
exten = _1NXXNXX,n,Goto(outbound-US,${EXTEN},1)
This is working perfectly fine (numbers changed to protect the
innocent!) until someone forwards their phone. When someone calls in
for them, the forwarded call becomes an outbound call and we are
overwriting the callerID rather than showing the original callerID.
Is there some way that I'm missing to distinguish between an outbound
call and a forwarded outbound call? Is there a better way to do what we
are doing? Thanks - John
We solved this with the following logic assuming outside callerIDs were
at least 7 digits long and internal extensions were less than 7 digits
(numbers changed to protect the real numbers):
exten = _1NXXNXX,1,ExecIf($[${CALLERID(num)} - 100
0]?Set(CALLERID(num)=6715728792))
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] Detecting Transfer
On Tue, 2009-09-15 at 16:44 -0500, Brent Davidson wrote:
Is there a way to detect if a call is a transfer in the dialplan? Here
is my issue: I have an office with 2 extensions. Under normal
circumstances any call that comes in should ring both extensions. I
accomplish this through a queue. The problem is that if the call is
answered on say extension 11 and the answerer wants to transfer the call
to the other phone, extension 10, transferring the call to extension 10
puts it back in the queue that again rings both phones. I want to set
the system up so that if the call is a transfer from the other extension
it will only ring the phone it's being transferred to. This is what I'm
currently doing (using AEL dialplan):
10 = {
if (${CALLERID(num)} = 11) {
internal-ext(${EXTEN},SIP/${EXTEN});
} else {
Queue (operator|tTnHr|||30);
}
Voicemail(1...@internal|u);
Hangup;
}
11 = {
if (${CALLERID(num)} = 10) {
internal-ext(${EXTEN},SIP/${EXTEN});
} else {
Queue (operator|tTnHr|||30);
}
Voicemail(1...@internal|u);
Hangup;
}
My only problem is that we have some extension duplication at other
offices and it is possible for an extension to come in from another
office with the same CallerID Number.
Is there a better way to do this?
snip
We did something very similar with a ring group. When a call comes in
to the main number, we have it dial ${ALLPHONES} which is
SIP/blahSIP/bleeSIP/bloo. When blee transfers to blah, it works just
as expected. Hope this helps - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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[asterisk-users] The o dial option
Hello, all. I see there is an o option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because of the default use of callerid. However I'm assuming it was changed to the current behavior for a good reason. Before we revert to the old behavior, I'd like to ask, why was it changed? What problems arose from the old behavior that provoked the change? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Duplicate DTMF
On Thu, 2009-09-10 at 15:26 -0400, Kristian Kielhofner wrote: On Wed, Sep 9, 2009 at 10:22 PM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several invalid extension or password incorrect messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might dial extension 1234, see 1234 on the phone from which we dialed, but see 112334 on the Asterisk console. We have seen this from cell phones calling via the PSTN (we use a SIP trunking carrier and do not handle the PSTN interface ourselves); we've seen it from land line phones via the PSTN, and have even seen it internally from our own Snom SIP phones. dtmfmode=auto but we have also tried setting it to rfc2833 and we have tried relaxdtmf set to both yes and no. We are running Asterisk 1.6.1.6 on CentOS 5.3. We really don't know what more to do to fix it. Googling shows that others have had this problem but I haven't seen a clear resolution other than playing with relaxdtmf. How do we solve this problem? Thanks - John Fairly typical for most SIP carriers... My blog entry may be able to illuminate this a bit: http://blog.krisk.org/2009/02/update-youve-been-waiting-for.html In short RFC2833 DTMF issues are fairly common. It's troublesome enough when trying to go directly to the Tier 1 carriers themselves. More than likely you're dealing with a reseller (carrier) that most likely inherits issues from their carrier and adds their own. A couple of weeks ago someone e-mailed me asking for RFC2833 assistance with Asterisk and a carrier using Sonus. Turns out: a) The carrier was a reseller of various other carriers that use Sonus. b) The carrier proxied RTP (and therefor RFC2833 events) through an Asterisk 1.2 machine; further mangling the RFC2833 events. Other than some drastic changes at the carrier there wasn't much that could be done... Sorry I can't offer more specific advice to your situation bit without an RTP packet capture there isn't much I (we) can do. P.S. - Ignore any suggestions for gain, etc. These are for Zap channels and do not apply to sip. Changing anything in zapata.conf will not affect your situation. I'm not even sure of the existence or purpose of relaxdtmf in sip.conf in Asterisk 1.4 or later. This may indeed be the case. I hesitated to ask our carrier (with whom we are quite happy thus far) since I believe we have seen this issue on internal calls (but only once as opposed to the consistent problem with external calls). I did anyway and they put us on a different switch. That seems to have solved the problem. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Duplicate DTMF
Hello, all. I've come across a nasty problem just as we are ready to put our first system into production. During our final testing, we were plagued with several invalid extension or password incorrect messages when we knew the information entered was correct. Upon investigation, we saw that DTMF signals were occasionally but not consistently duplicated. We might dial extension 1234, see 1234 on the phone from which we dialed, but see 112334 on the Asterisk console. We have seen this from cell phones calling via the PSTN (we use a SIP trunking carrier and do not handle the PSTN interface ourselves); we've seen it from land line phones via the PSTN, and have even seen it internally from our own Snom SIP phones. dtmfmode=auto but we have also tried setting it to rfc2833 and we have tried relaxdtmf set to both yes and no. We are running Asterisk 1.6.1.6 on CentOS 5.3. We really don't know what more to do to fix it. Googling shows that others have had this problem but I haven't seen a clear resolution other than playing with relaxdtmf. How do we solve this problem? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Recommendations about infrastructure to use with Asterisk
to attack. Oh, I should mention we use the ISCS project (http://iscs.sourceforge.net) to restrict all network traffic on an as-needed basis - firepiping instead of firewalling - hence the concern that we want to restrict network access even on the internal networks. As a result, hard phones are canreinvite=nonat and have UDP high ports open whereas soft phones are canreinvite=no and do not allow access to UDP high ports unless associated with an initial SIP conversation. This is certainly a big topic but I hope this gives you some place to start. It is what we did and we are please so far with our results in test. We will stress test this implementation in production this coming week. Good luck - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Allowing multiple callers to join a public speaking session...?
On Wed, 2009-09-02 at 14:03 -0400, li...@mgreg.com wrote: On Sep 2, 2009, at 1:33 PM, Jeff LaCoursiere wrote: Hi Michael, Yes, I think you are on the right track. A Meetme conference is what you need, and perhaps a service to provide a DID number that would allow multiple people to call in to your conference at the same time (without purchasing POTS hardware, dealing with echo issues, etc.). Checkout www.ipcomms.net. I use them for a number of DID services. Their rates are decent and their support folks know asterisk. Cheers, j Thanks for the posts thus far! In all honesty I'm looking for a complete in house solution. I don't mind spending up to $500-600 on equipment if necessary. I just want to know that when I'm done there are no residual costs, etc. Is Asterisk capable of this kind of setup/ management? As for labor, I'm willing to donate as much as is necessary. snip Absolutely. It doesn't sound like you need much firepower. You may even be able to carve off a virtual server for it. We don't do that in order to minimize latency but I'm sure lots of folks swear by such a setup. You will have the typical maintenance - updates, security patches, any client side changes. I would imagine your biggest challenge will be getting people into the system. If they are all internal (I was originally assuming they were not), they can all use soft phones and head sets. Since it is a monologue, you may even be able to dispense with the headsets. If folks are calling in from outside your network, it gets a little trickier. If they all have Internet connections, they can establish direct SIP connections to your PBX. If they are coming in from the PSTN, you will need phone lines. You could talk to a VoIP carrier and see if they can replace your PSTN access and then you would have the best of all worlds. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] outbound calls not ringing still
On Wed, 2009-09-02 at 21:31 +, Ott Rose wrote: i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133...@216.82.224.202 SIP/2.0 Via: SIP/2.0/UDP 216.82.224.202:5060;branch=z9hG4bK3691b08c;rport From:8592192438sip:8592192...@64.191.130.78;tag=as0707d433 To:sip:+185993133...@216.82.224.202 Contact:sip:8592192...@216.82.224.202 Call-ID: 0f3bdcc9171ef53148e7bab413aea...@64.191.130.78 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 02 Sep 2009 21:10:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:103 h263-1998/9 a=sendrecv snip I know very little about how ringing works but are they providing any kind of status information to you? Do you need to furnish the ring if they are not? It seems to me I saw quite a few articles about providing ring tone, what causes it to fail, and how to work around it. I assume you've searched for those already. Just a few thoughts - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Selective canreinvite in multi-tenant environment
On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote: Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, we can't, but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic associated with intra-tenant traffic from the Asterisk server and reduce the intra-tenant latency by doing so. However, I am very, very hesitant to allow our VPN connections to tenants to function as a router between tenants allowing one tenant to directly access phones on another tenant (that's not as wild as it sounds because of our use of the ISCS project - iscs.sourceforge.net). Since the tenants are all connecting via VPN, we are using RFC1918 addresses and no NAT is involved thus the canreinvite=nonat option does not help us. If we set canreinvite=nonat, that will allow for intra-tenant direct media but, if one tenant tries to call another via SIP, it will redirect the media at the Asterisk level but the packets will be dropped at the firewall / router level (or sooner as there may be no route to the destination) and the call will connect but with no sound. Any guidance would be greatly appreciated. Thanks - John As mentioned in another post, we were able to solve this by setting a w dial option to all inbound SIP calls from the Internet. Thus, all internal calls could reinvite but external calls could not. However, just when we thought this was working splendidly well, we turned up another roadblock - transfers. We find that once we transfer a call using our Snom phones, the call between the new call partners does not seem bound by the w constraint, Asterisk tries to reinvite the call, and the audio breaks because the firewall cannot associate the new RTP stream with a SIP session. How have others gotten around the problem of transfers causing reinvites on calls which should not allow reinvites? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Selective canreinvite in multi-tenant environment
Hello, all. In our multi-tenant environment, we would like to be able to use the reinvite media redirection within Asterisk for calls within a tenant but not between tenants. We would like inter-tenant calls to be fully proxied by the Asterisk server. I think the answer is, we can't, but I thought I'd ask anyway. I'd dearly like to remove the substantial traffic associated with intra-tenant traffic from the Asterisk server and reduce the intra-tenant latency by doing so. However, I am very, very hesitant to allow our VPN connections to tenants to function as a router between tenants allowing one tenant to directly access phones on another tenant (that's not as wild as it sounds because of our use of the ISCS project - iscs.sourceforge.net). Since the tenants are all connecting via VPN, we are using RFC1918 addresses and no NAT is involved thus the canreinvite=nonat option does not help us. If we set canreinvite=nonat, that will allow for intra-tenant direct media but, if one tenant tries to call another via SIP, it will redirect the media at the Asterisk level but the packets will be dropped at the firewall / router level (or sooner as there may be no route to the destination) and the call will connect but with no sound. Any guidance would be greatly appreciated. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Selective canreinvite in multi-tenant environment
On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote:
Hello, all. In our multi-tenant environment, we would like to be able
to use the reinvite media redirection within Asterisk for calls within a
tenant but not between tenants. We would like inter-tenant calls to be
fully proxied by the Asterisk server. I think the answer is, we
can't, but I thought I'd ask anyway.
I'd dearly like to remove the substantial traffic associated with
intra-tenant traffic from the Asterisk server and reduce the
intra-tenant latency by doing so. However, I am very, very hesitant to
allow our VPN connections to tenants to function as a router between
tenants allowing one tenant to directly access phones on another tenant
(that's not as wild as it sounds because of our use of the ISCS project
- iscs.sourceforge.net).
Since the tenants are all connecting via VPN, we are using RFC1918
addresses and no NAT is involved thus the canreinvite=nonat option does
not help us. If we set canreinvite=nonat, that will allow for
intra-tenant direct media but, if one tenant tries to call another via
SIP, it will redirect the media at the Asterisk level but the packets
will be dropped at the firewall / router level (or sooner as there may
be no route to the destination) and the call will connect but with no
sound.
Any guidance would be greatly appreciated. Thanks - John
Ah, I found a way - that'll teach me to say there's something one can't
do in Asterisk! However, it has turned up some unexpected behavior.
First the new problem and then the original solution.
The new problem: The documentation says the L() option prevents
reinvites. However, this does not appear to be true. When we dial a
SIP device with canreinvite=nonat using L(3060) (8.5 hours), we see
the RTP traffic shunted to be directly between the end points. If we
use something like t instead, the reinvite is not issued and traffic is
handled B2BUA.
The original solution which works well except for the above problem:
I'll explain this at two levels for the sake of those who do not need
all the details. The inter-tenant calls are done via sip URLs which are
handled differently from the intra-tenant, i.e., within a tenant one
might dial 333 to get Jane whereas, from a different tenant, one would
dial sip:j...@mycompany.com. The inbound uri handler simply redirects
the dial plan to the dial plan for extension 333. This in turn calls
our common dialing macro.
We realized we could set a persistent channel variable in the uri
handler, e.g., __DOPTS, set it to some dialing option which prevents
reinvites, and then goto the extension part of the plan.
We've relegated most of the functions handled by these options to the
phones, e.g., transfer, hangup, record, so we didn't want to duplicate
functionality. For that reason, we elected to use the L() option except
it does not seem to work as mentioned above :-(
Here are the gory details for those who actually need this functionality
rather than those just curious.
There are several issues involved. They are mainly provoked by the
capabilities or lack thereof of the firewalls. Some of our clients have
firewalls which are SIP aware and can handle the RTP port assignment as
part of the SIP packet flow. Others are not SIP aware and thus must
allow high UDP ports to be open for the RTP traffic.
For those with SIP aware firewalls, we do not really need this fix.
Allowing inter-tenant traffic is no more dangerous than allowing inbound
SIP calls to phones since we can restrict it to the SIP port. Of
course, even that is too insecure for many installations.
For those without SIP aware firewalls, we have two issues. The first is
security - we do not want to open a raft of high UDP ports for RTP
traffic which could possible be exploited for other purposes. The
second is NAT. If they attempted a direct RTP conversation rather than
passing through the NAT aware Asterisk setup, the RTP traffic will not
be properly NAT'd. Hmm . . . on second thought, that is probably not
true as I do not believe the RTP traffic contains embedded IP address
information in the data portion of the packet and the SIP signalling is
still being handled by Asterisk. I'll have to play with that one.
In any event, we have to worry about both inbound and outbound calls.
To prevent inbound calls from being reinvited, we added the dial option
to kill reinvites (which we set in a global macro named KREINVITE) to
the inbound sip url handler.
As one would imagine, to prevent outbound SIP calls from being
reinvited, we added the ${KREINVITE} option to the outbound SIP URL
handler.
The final result looks something like this:
KREINVITE=L(3060) ; used to kill reinvite by inserting as an option in the
dial command - timeout after 8.5 hours (30,600,00 ms)
[macro-common]
; ARG1 = extension to dial
; ARG2 = extension for voice mail
; ARG3 = context for voice mail
; ARG4 = extension for followme (optional)
; ARG5 = Timeout
Re: [asterisk-users] Selective canreinvite in multi-tenant environment
On Thu, 2009-08-27 at 16:20 -0500, Kevin P. Fleming wrote: John A. Sullivan III wrote: Hope this helps someone else. Improvements, suggestions, and constructive criticism welcome. If anyone knows why we are not getting the expected reinvite prevention from the L() option, please let me know. Thanks - John Umm... it's a feature? 'reinvite prevention' was never the purpose of the L() option, it was a side effect of its implementation. The code has been improved to allow the reinvite to occur in spite of the L() option being used, and then the media will be reinvited back to Asterisk when the time comes to play warnings, drop the call, etc. grin Glad to hear of the improvement - just sorry for us. We'll use the safest of the remaining options we can think of. Is there a better way to do what we are trying to do? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] netfilter conntrack mangling canreinvite?
On Tue, 2009-08-25 at 21:07 -0400, John A. Sullivan III wrote: Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to reinvite has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop another day into verifying this, may I ask if anyone else has had a similar problem and found a solution? It appears conntrack is rewriting the SDP so that the address is reverted to the PBX address. Here are the relevant SDP portion of a reinvite captured on the PBX using tcpdump and displayed in Wireshark. The PBX is at 172.x.x.8 and the phone is at 10.x.x.193: Owner/Creator, Session Id (o): root 1417450700 1417450701 IN IP4 10.x.x.183 Owner Address: 10.x.x.183 Connection Information (c): IN IP4 10.x.x.183 Connection Address: 10.x.x.183 Here is a similar sequence but captured from the phone itself: Owner/Creator, Session Id (o): root 595629021 595629022 IN IP4 172.x.x.8 Owner Address: 172.x.x.8 Connection Information (c): IN IP4 172.x.x.8 Connection Address: 172.x.x.8 It would appear conntrack is incorrectly fixed the packet. I noticed newer kernels have sip_direct_media and sip_direct_signalling options. I don't know if those apply but they do not seem to be present in our CentOS 5.3 kernel. I'll probably spend most of tomorrow confirming this hypothesis and investigating solutions so I'd be deeply appreciative for any time-saving advice. Thanks - John The ip_nat_sip conntrack module was indeed the culprit. Apparently this can be fixed in newer kernels by setting the sip_direct_media=0 option for ip_conntrack_sip in modprobe.conf. However, since our CentOS 5.3 version of the kernel does not support this, we disabled ip_nat_sip and returned responsibility for managing NAT to sip.conf. Hope this helps someone else - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] followme app
On Tue, 2009-08-25 at 16:28 +0200, harry R wrote:
Hi
Someone may give me an example of followme() application using in a
dialplan (including what to configure in followme.conf) ?
I use asterisk 1.6.1 so if your example can match to that release it's
will be wonderfull.
snip
We are using followme on 1.6.1 with the slight complication that we are
a multi-tenant environment. Since we have clients with the same
extension numbers living in different contexts, we cannot directly map
extensions to followme definitions. Here's how we handled it.
In followme.conf (actually in included files - one per client), we have
definitions using a globally unique identifier per user, e.g.,
[1234561] ; John Sullivan
context=zx400
number=12345678901,30
number=12345678902,36
The call to followme is initiated as part of our basic call handling
macro:
[macro-common]
; ARG1 = extension to dial
; ARG2 = extension for voice mail
; ARG3 = context for voice mail
; ARG4 = extension for followme (optional)
; ARG5 = Timeout is seconds until voice mail / followme (optional - defaults to
24)
exten = s,1,Set(TM=${IF(${ISNULL(${ARG5})}?24:${ARG5})})
exten = s,n,Dial(${ARG1},${TM})
exten = s,n,Wait(0.5)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${ISNULL(${ARG4})}]?vm)
exten = s-NOANSWER,n,background(custom/ImSorry)
exten = s-NOANSWER,n,background(custom/Silence-0.25)
exten = s-NOANSWER,n,background(custom/No1AtXten)
exten = s-NOANSWER,n,background(custom/Silence-0.5)
exten = s-NOANSWER,n,background(custom/Press${ARG2})
;exten = s-NOANSWER,n,background(custom/Silence-0.25)
exten = s-NOANSWER,n,background(custom/4VoiceMail)
exten = s-NOANSWER,n,background(custom/Silence-0.5)
exten = s-NOANSWER,n,background(custom/Press${ARG4})
exten = s-NOANSWER,n,background(custom/Silence-0.25)
exten = s-NOANSWER,n,background(custom/TryToFindPerson)
exten = s-NOANSWER,n,WaitExten(5)
exten = s-NOANSWER,n(vm),Voicemail(${macro_ext...@${arg3},u)
exten = s-BUSY,1,Voicemail(${macro_ext...@${arg3},b)
exten = _s-.,1,Goto(s-NOANSWER,1)
where they press some key for voice mail and another key to have the
system try to find the person, i.e., followme. This key choice is
then handled in the originating context:
exten = 2,1,GotoIf(${ISNULL(${FM})}?i,1)
exten = 2,n,FollowMe(${FM},san)
exten = 2,n,Goto(1,1)
where we first check to make sure it was a hand-off from the macro and
not a misdialed, invalid extension.
FM is a variable set to track the globally unique followme identifier.
Here is a sample user's extension definition showing how we set the
identifier:
; John Sullivan
exten = xxx,1,Set(__VM=312) ; VoiceMail ID
exten = xxx,n,Set(__FM=11) ; Followme ID
exten = xxx,n,Macro(common,SIP/jasiii,1,zx400,2)
exten = 8xxx,1,VoiceMail(x...@zx400)
exten = 7xxx,1,VoiceMailMain(x...@zx400)
exten = 7xxx,n,Hangup()
I realize that's a somewhat complicated example and, as we are very new
to asterisk, any critiques and improvements are welcome. In summary:
1. We defined a followme for a user with a unique ID - in many
cases this can be the extension
2. Our busy / not-available routine offers an option for followme,
traps this number as an extension just like an automated
attendant
3. Checks to make sure that extension was not an accident, e.g., if
the number pressed for followme is 2, we don't want people
landing in followme because they dialed 2 from the main
auto-attendant if they happen to be in the same context
4. Call followme by passing it the unique ID and any options
Hope this helps. Again, bear in mind that we are new to this so if
someone suggests a better way, they are probably right :-) - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 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] followme app
On Tue, 2009-08-25 at 12:21 -0400, John A. Sullivan III wrote:
On Tue, 2009-08-25 at 16:28 +0200, harry R wrote:
Hi
Someone may give me an example of followme() application using in a
dialplan (including what to configure in followme.conf) ?
I use asterisk 1.6.1 so if your example can match to that release it's
will be wonderfull.
snip
We are using followme on 1.6.1 with the slight complication that we are
a multi-tenant environment. Since we have clients with the same
extension numbers living in different contexts, we cannot directly map
extensions to followme definitions. Here's how we handled it.
In followme.conf (actually in included files - one per client), we have
definitions using a globally unique identifier per user, e.g.,
[1234561] ; John Sullivan
context=zx400
number=12345678901,30
number=12345678902,36
The call to followme is initiated as part of our basic call handling
macro:
[macro-common]
; ARG1 = extension to dial
; ARG2 = extension for voice mail
; ARG3 = context for voice mail
; ARG4 = extension for followme (optional)
; ARG5 = Timeout is seconds until voice mail / followme (optional - defaults
to 24)
exten = s,1,Set(TM=${IF(${ISNULL(${ARG5})}?24:${ARG5})})
exten = s,n,Dial(${ARG1},${TM})
exten = s,n,Wait(0.5)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,GotoIf($[${ISNULL(${ARG4})}]?vm)
exten = s-NOANSWER,n,background(custom/ImSorry)
exten = s-NOANSWER,n,background(custom/Silence-0.25)
exten = s-NOANSWER,n,background(custom/No1AtXten)
exten = s-NOANSWER,n,background(custom/Silence-0.5)
exten = s-NOANSWER,n,background(custom/Press${ARG2})
;exten = s-NOANSWER,n,background(custom/Silence-0.25)
exten = s-NOANSWER,n,background(custom/4VoiceMail)
exten = s-NOANSWER,n,background(custom/Silence-0.5)
exten = s-NOANSWER,n,background(custom/Press${ARG4})
exten = s-NOANSWER,n,background(custom/Silence-0.25)
exten = s-NOANSWER,n,background(custom/TryToFindPerson)
exten = s-NOANSWER,n,WaitExten(5)
exten = s-NOANSWER,n(vm),Voicemail(${macro_ext...@${arg3},u)
exten = s-BUSY,1,Voicemail(${macro_ext...@${arg3},b)
exten = _s-.,1,Goto(s-NOANSWER,1)
where they press some key for voice mail and another key to have the
system try to find the person, i.e., followme. This key choice is
then handled in the originating context:
exten = 2,1,GotoIf(${ISNULL(${FM})}?i,1)
exten = 2,n,FollowMe(${FM},san)
exten = 2,n,Goto(1,1)
where we first check to make sure it was a hand-off from the macro and
not a misdialed, invalid extension.
FM is a variable set to track the globally unique followme identifier.
Here is a sample user's extension definition showing how we set the
identifier:
; John Sullivan
exten = xxx,1,Set(__VM=312) ; VoiceMail ID
exten = xxx,n,Set(__FM=11) ; Followme ID
exten = xxx,n,Macro(common,SIP/jasiii,1,zx400,2)
exten = 8xxx,1,VoiceMail(x...@zx400)
exten = 7xxx,1,VoiceMailMain(x...@zx400)
exten = 7xxx,n,Hangup()
I realize that's a somewhat complicated example and, as we are very new
to asterisk, any critiques and improvements are welcome. In summary:
1. We defined a followme for a user with a unique ID - in many
cases this can be the extension
2. Our busy / not-available routine offers an option for followme,
traps this number as an extension just like an automated
attendant
3. Checks to make sure that extension was not an accident, e.g., if
the number pressed for followme is 2, we don't want people
landing in followme because they dialed 2 from the main
auto-attendant if they happen to be in the same context
4. Call followme by passing it the unique ID and any options
Hope this helps. Again, bear in mind that we are new to this so if
someone suggests a better way, they are probably right :-) - John
snip
Oops! I didn't consistently expunge all the internal data. The __FM=
should be 1234561 and the __VM= should be xxx. That will make the above
example internally consistent. Sorry - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Breaking news, but what happened? 11.000 channels on one server
On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote: 25 aug 2009 kl. 16.20 skrev Olivier: I would be curious to know if bonding 2 Ethernet ports together would help to push the upper limit a bit further ... (by the way, this limit is 11000 channels or 5500 calls, isn't it ? Yes, this is 11.000 channels. Bonding is good advice, provided we have a switch that can handle that. Gotta find a place to borrow such a switch. /Olle You don't necessarily need a switch to support it. One can use alb mode in Linux on any old switch and it works reasonably well other than for some excessive ARP traffic. However, as we found out the hard way when building our Nexenta SAN, bonding works very well with many-to-many traffic but does very little to boost one-to-one network flows. They will all collapse to the same pair of NICs in most scenarios and, in the one mode where they do not, packet sequencing issues will reduce the bandwidth to much less than the sum of the connections. Take care - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Breaking news, but what happened? 11.000 channels on one server
On Wed, 2009-08-26 at 06:18 +1000, Alex Samad wrote: On Tue, Aug 25, 2009 at 07:30:08PM +0200, Olle E. Johansson wrote: 25 aug 2009 kl. 18.50 skrev John A. Sullivan III: On Tue, 2009-08-25 at 18:28 +0200, Olle E. Johansson wrote: 25 aug 2009 kl. 16.20 skrev Olivier: [snip] mode in Linux on any old switch and it works reasonably well other than for some excessive ARP traffic. However, as we found out the hard way when building our Nexenta SAN, bonding works very well with many-to-many traffic but does very little to boost one-to-one network flows. They will all collapse to the same pair of NICs in most scenarios and, in the one mode where they do not, packet sequencing issues will reduce the bandwidth to much less than the sum of the connections. Take care - John That is very good feedback - thanks, John! Which means that my plan B needs to be put in action. Well, I did create a new branch for it yesterday... ;-) any thoughts of different media like 10G ethernet or infiniband ? snip Yes, this is drifting a little off-topic but good network design does provide the foundation for good Asterisk design. If we have lots of servers talking to lots of servers, bonding over Gig links works very well. But as we build fewer very big servers via virtualization or, as in this case, trying to make a single large server do the work previously handled by several, the network bandwidth becomes a huge issue. Because almost all bonding algorithms choose a single path for a flow of data (usually based upon MAC address but sometimes on IP address or even socket), bonding becomes less useful in these scenarios. In fact, it is even worse - even in cases where the OS stack (e.g., Linux) will support bonding based upon data above the MAC layer, the switches frequently do not and will again collapse several paths into one as soon as the data crosses the switch. Thus, for few-to-few traffic patterns, bigger pipes such as 10G are better than bonded pipes. Specifically, 10 bonded 1 Gbps links will effectively yield 1 Gbps throughput as opposed to 1 10Gbps link yielding 10 Gpbs throughput. As an aside, in our iSCSI work, we found latency to be a huge issue if the file block size was small (e.g., Linux files - 4K block size). Thus, the lower latency of faster protocols is a huge performance booster. This will not be so much of an issue with Asterisk where the difference between 100 usecs and 10 usecs in negligible. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] netfilter conntrack mangling canreinvite?
Hello, all. Since implementing an iptables firewall between the Asterisk PBX and several SIP phones, the Asterisk PBX ability to reinvite has been broken even when the phones are on the same network (i.e., no firewall between the phones). We've been beating our heads against the wall thinking it was the complex rule set but it appears the issue is ip_conntrack_sip. Before I drop another day into verifying this, may I ask if anyone else has had a similar problem and found a solution? It appears conntrack is rewriting the SDP so that the address is reverted to the PBX address. Here are the relevant SDP portion of a reinvite captured on the PBX using tcpdump and displayed in Wireshark. The PBX is at 172.x.x.8 and the phone is at 10.x.x.193: Owner/Creator, Session Id (o): root 1417450700 1417450701 IN IP4 10.x.x.183 Owner Address: 10.x.x.183 Connection Information (c): IN IP4 10.x.x.183 Connection Address: 10.x.x.183 Here is a similar sequence but captured from the phone itself: Owner/Creator, Session Id (o): root 595629021 595629022 IN IP4 172.x.x.8 Owner Address: 172.x.x.8 Connection Information (c): IN IP4 172.x.x.8 Connection Address: 172.x.x.8 It would appear conntrack is incorrectly fixed the packet. I noticed newer kernels have sip_direct_media and sip_direct_signalling options. I don't know if those apply but they do not seem to be present in our CentOS 5.3 kernel. I'll probably spend most of tomorrow confirming this hypothesis and investigating solutions so I'd be deeply appreciative for any time-saving advice. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Breaking news, but what happened? 11.000 channels on one server
On Tue, 2009-08-25 at 21:57 -0400, David Backeberg wrote: On Tue, Aug 25, 2009 at 12:50 PM, John A. Sullivan IIIjsulli...@opensourcedevel.com wrote: You don't necessarily need a switch to support it. One can use alb mode in Linux on any old switch and it works reasonably well other than for some excessive ARP traffic. However, as we found out the hard way when building our Nexenta SAN, bonding works very well with many-to-many traffic but does very little to boost one-to-one network flows. They will all collapse to the same pair of NICs in most scenarios and, in the one mode where they do not, packet sequencing issues will reduce the bandwidth to much less than the sum of the connections. Take care - Your claims make sense for a typical Machine A has one IP address Machine B has one IP address And there is only one route between A and B. In this scenario, yes, all calls take same route. But what about giving each machine two addresses, two routes. And halve your calls between the two paths between the same systems. Doesn't this get around your problem, and allow you a chance to saturate double the number of interfaces? If you have four interfaces (as my new boxes do), lather, rinse, repeat. Anybody have any reason why spreading the bandwidth across multiple routes wouldn't get around this problem? snip Yes, that's correct and exactly what we did in our SAN environment. There are some issues. You will generally not want them all on the same IP network - the inbound traffic may spread across the four addresses if told to do so but the reply traffic will likely go out the default interface. If they are four distinct IP networks, it means dividing the end users among the multiple networks. In the case of our SAN, we did it without a router using logical networks on the same physical medium. With iproute2 and secondary routing tables, one can be even more creative. In fact, having many phones going to one Asterisk device will probably work well with bonding because it is many to one and each combination of MAC addresses will be treated as a different traffic stream. However, if I recall, the testing environment was two or three asterisk systems talking to each other, wasn't it? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] MEETME how to lock the conference if no admin are connected
On Wed, 2009-08-19 at 09:16 +0200, BERGANZ François wrote:
hello
is it possible to lock a conference IF no admin are connected ?
or how to do to have a conference offline?
snip
If I understand you correctly, we are doing something similar. When
users call into a conference, they hear music on hold and cannot speak
to each other until the moderator joins the conference. Our calls to
meetme are via macros but they should give you the idea:
[macro-confmod] ;conference moderator
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cMaAsx)
[macro-confpart] ;conference participant
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cIMswx,${ARG2})
I believe the critical options are the w for the participant (regular
users) which says wait until a marked user joins the conference and
the A for the moderator which designates the moderator as such a
marked user.
I don't understand what you mean by an offline conference. Hope this
helps - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] outbound calls not ringing
On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] outbound calls not ringing
sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 __ Hotmail® is up to 70% faster. Now good news travels really fast. Try it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] outbound calls not ringing
Oops! - You're using FreePBX - someone who knows more about FreePBX will have to help you as I don't. May I also suggest that you bottom post in future responses rather than top post; that makes it a little easier to follow. Good luck - John On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote: here is my sip.conf. i don't see it. ;; ; 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 ; ;; ; [general] ; These files will all be included in the [general] context ; #include sip_general_additional.conf ;sip_general_custom.conf is the proper file location for placing any sip general ;options that you might need set. For example: enable and force the sip jitterbuffer. ;If these settings are desired they should be set the sip_general_custom.conf file. ; ; jbenable=yes ; jbforce=yes ; ;It is also the proper place to add the lines needed for sip nat'ing when going ;through a firewall. For nat'ing you'd need to add the following lines: ; nat=yes , externip= , localhost= , and optionally fromdomain= . ; #include sip_general_custom.conf ;sip_nat.conf is here for legacy support reasons and for those that upgrade ;from previous versions. If you have this file with lines in it please make ;sure they are not duplicated in sip_general_custom.conf, if so remove them ;from sip_nat.conf as sip_general_custom.conf will have precedence. #include sip_nat.conf ;sip_registrations_custom.conf is for any customizations you might need to do to ;the automatically generated registrations that FreePBX makes. ; #include sip_registrations_custom.conf #include sip_registrations.conf ; These files should all be expected to come after the [general] context ; #include sip_custom.conf #include sip_additional.conf ;sip_custom_post.conf If you have extra parameters that are needed for a ;extension to work to for example, those go here. So you have extension ;1000 defined in your system you start by creating a line [1000](+) in this ;file. Then on the next line add the extra parameter that is needed. ;When the sip.conf is loaded it will append your additions to the end of ;that extension. ; #include sip_custom_post.conf From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 12:17:15 -0400 Subject: Re: [asterisk-users] outbound calls not ringing sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 __ Hotmail® is up to 70% faster. Now good news travels really fast. Try it now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Moderator access to meetme allowed despite pin
Hello, all. I've solved my own problem but will post it here in case
someone else has the same misunderstanding in the future.
We thought we had set up our meetme so that regular users entered the
conference without a pin but could not speak to each other until the
moderator arrived. We enforced pin entry on the moderator . . . or at
least so we thought. If the moderator waited long enough without
entering a pin, they were entered into the conference.
The problem was we had misinterpreted the meetme.conf and
extensions.conf pin parameters and had understood them essentially
backwards.
We setup our meetme.conf file something like this:
conf = 100,,321
conf = 102,,432
erroneously assuming this meant regular users did not require a pin but
moderators did.
Our meetme application was called via macros as follows:
[macro-confmod] ;conference moderator
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cMaAsx)
[macro-confpart] ;conference participant
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cIMswx)
and our extensions like this:
exten = 6151,1,Macro(confpart,100)
exten = 5151,1,Macro(confmod,100)
We did not enter a pin in the extension because we thought that meant a
pin was required.
Now we realize what this really said was, If the moderator enters the
moderator pin, make them the moderator, otherwise, let them and everyone
else in without a pin.
So our first misunderstanding was that an empty user pin and populated
moderator pin meant only the moderator was required to enter a pin. The
second was that placing the pin in the extension meant we were requiring
the pin when, actually it is doing the opposite - it is providing the
pin so the user does not have to.
To do what we originally intended, our meetme.conf should have required
a pin for everyone with a different pin for the moderator like so:
conf = 100,123,321
conf = 102,234,432
We then should have populated the pin in the extension for the regular
users but not for the moderator like so:
[macro-confmod] ;conference moderator
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cMaAsx)
[macro-confpart] ;conference participant
exten = s,1,Macro(conference,${MACRO_EXTEN})
exten = s,n,MeetMe(${ARG1},cIMswx, ${ARG2})
exten = 6151,1,Macro(confpart,100,123)
exten = 5151,1,Macro(confmod,100)
Hope this keeps someone else from accidentally opening their
conferencing system to the world! - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] sflphone questions
On Mon, 2009-08-10 at 21:37 -0500, Tom Poe wrote: I want to set sflphone as extension on asterisk. I have a sip account/DID with vitelity.net. Not sure what to put in the wizard: alias ??? hostname ??? is this the asterisk server hostname, or the hostname where my sflphone is sitting on the lan (it's a home network) username ??? is this the assigned extension number? password ??? is this the assigned extension number password? snip I've never used sflphone and have been reasonably happy with Twinkle but it looks like an interesting alternative. My guess is the alias may be a callerID, the hostname is the asterisk server, the username is the SIP ID, and the password is the SIP secret. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement
That's interesting. I was always under the impression from what I read that T.38 was an unreliable, experimental crap-shoot at best and something that should be avoided for production systems - that the only reliable solution for FAX was still PSTN lines. Is this no longer true and all the dire caveats about T.38 faxing obsolete? This is quite important to us as we are planning to launch a FAX service for our clients shortly and are dreading a return to banks of modems to handle analog lines. Thanks - John On Mon, 2009-08-03 at 14:32 +1000, Klaverstyn, David C wrote: Faxing over SIP never worked for me. The faxes would always fail. When I saw the information about T.38, I decided to immediately upgrade to 1.6.0.11-rc2 from 1.6.0.10 and try it. I was amazed. Without having to change anything in my configuration faxes just worked. I have tested it with multiple faxes, short and long, and faxes with images and they all came through. Well done guys. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Team Sent: Monday, 3 August 2009 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1,and 1.6.2.0-beta4 Release Announcement The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The related security advisory AST-2009-004 has been released along with this announcement. Please read that advisory for more information. The release candidates and betas, in addition to other fixes, contain a major re-work of the T.38 support in Asterisk. If you've been having trouble with T.38 in the 1.6 series, you are strongly encouraged to try one of these release candidates to determine if these changes fixed your T.38 issues. For a full list of changes in these releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.0.11-rc2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.1.2 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.1.3-rc1 http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog- 1.6.2.0-beta4 Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Upgrading from 1.6.1.1 to 1.6.1.2
Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Upgrading from 1.6.1.1 to 1.6.1.2
On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote: Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John We've hit a problem even before installing. We're using Zimbra as IMAP storage for our voicemails. When we run make menuselect in 1.6.1.2, the IMAP storage option is disabled (XXX). When we run menuselect in 1.6.1.1 on the same system, it is available and enabled. Did we miss something or is this a bug? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] SIP AND NAT
On Mon, 2009-08-03 at 13:29 -0400, Ketema Harris wrote: I recently did a set up where I replaced a simple D-link home router that was having trouble processing a T1's worth of bandwidth with a linux machine running iptables. the kernel was 2.6.29-r5 and I chose the SIP connection tracking modules from the menuconfig. Router worked fine for normal traffic, but I was unable to get the SIP phones to work. Using ngrep it was plain to see that the although the packets going out were reaching their destination the data inside the sip headers all contained non routable IPs. I used lsmod and saw that the following modules: nf_nat_sip 5084 0 nf_nat 16400 3 nf_nat_sip,ipt_MASQUERADE,iptable_nat nf_conntrack_ipv4 11912 3 iptable_nat,nf_nat nf_defrag_ipv4 1788 1 nf_conntrack_ipv4 were loaded. I also googled and found the http://www.iptel.org/ sipalg/ website, but since this seemed to be a little dated I assumed the modules contained in the kernel source tree were newer and more reliable my questions are: What is the correct way(or resource to find a way) to get a linux firewall to work with SIP so that the NAT issue is not an issue ? snip Not an area of great expertise for me. I would think nf_nat_sip would take care of it but I'm surprised to not see conntrack_sip. Here is what is running on our firewall (not that we do a lot with NAT'd sip but the little we've done seems to work): [r...@fw01 ~]# lsmod | grep sip ip_nat_sip 37313 0 ip_conntrack_sip 41745 1 ip_nat_sip ip_nat 52845 5 ip_nat_h323,ip_nat_irc,ip_nat_ftp,ip_nat_sip,iptable_nat ip_conntrack 91237 13 ip_nat_h323,ip_nat_irc,ip_nat_ftp,ip_nat_sip,ip_conntrack_tftp,ip_conntrack_irc,ip_conntrack_h323,ip_conntrack_ftp,ip_conntrack_sip,ip_conntrack_netbios_ns,xt_state,iptable_nat,ip_nat -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Upgrading from 1.6.1.1 to 1.6.1.2
On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote: On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote: On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote: Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John We've hit a problem even before installing. We're using Zimbra as IMAP storage for our voicemails. When we run make menuselect in 1.6.1.2, the IMAP storage option is disabled (XXX). When we run menuselect in 1.6.1.1 on the same system, it is available and enabled. Did we miss something or is this a bug? Thanks - John Re-run configure on 1.6.1.1. It's likely that the option will go away, as the dependency is no longer met. I'm not sure I understand. Nothing has changed to make the dependency fail. This is the same device where we are quite successfully running voicemail in IMAP - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Upgrading from 1.6.1.1 to 1.6.1.2
On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote: On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote: On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote: On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote: Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John We've hit a problem even before installing. We're using Zimbra as IMAP storage for our voicemails. When we run make menuselect in 1.6.1.2, the IMAP storage option is disabled (XXX). When we run menuselect in 1.6.1.1 on the same system, it is available and enabled. Did we miss something or is this a bug? Thanks - John Re-run configure on 1.6.1.1. It's likely that the option will go away, as the dependency is no longer met. I'm not sure I understand. Nothing has changed to make the dependency fail. This is the same device where we are quite successfully running voicemail in IMAP - John Very strange, I did as you suggested and sure enough, IMAP is disabled as an option in 1.6.1.1. I'll have to dig a little deeper as I believe the most we may have done was a yum update! Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Upgrading from 1.6.1.1 to 1.6.1.2
On Mon, 2009-08-03 at 14:27 -0400, John A. Sullivan III wrote: On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote: On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote: On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote: On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote: Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John We've hit a problem even before installing. We're using Zimbra as IMAP storage for our voicemails. When we run make menuselect in 1.6.1.2, the IMAP storage option is disabled (XXX). When we run menuselect in 1.6.1.1 on the same system, it is available and enabled. Did we miss something or is this a bug? Thanks - John Re-run configure on 1.6.1.1. It's likely that the option will go away, as the dependency is no longer met. I'm not sure I understand. Nothing has changed to make the dependency fail. This is the same device where we are quite successfully running voicemail in IMAP - John Very strange, I did as you suggested and sure enough, IMAP is disabled as an option in 1.6.1.1. I'll have to dig a little deeper as I believe the most we may have done was a yum update! Thanks - John Ah, I remember now and shame on us for not documenting it. I'll record it here in case someone else hits the same thing. The version of libc-client that ships with CentOS 5.3 is too old to work with Zimbra. We thus needed to use the later imap-2007e version. configure needs to point to it, in our case: ./configure --with-imap=/home/compuser/Asterisk/imap-2007e I'll sheepishly add that to our internal documentation now :-( -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Upgrading from 1.6.1.1 to 1.6.1.2
On Mon, 2009-08-03 at 14:52 -0400, John A. Sullivan III wrote: On Mon, 2009-08-03 at 14:27 -0400, John A. Sullivan III wrote: On Mon, 2009-08-03 at 13:49 -0400, John A. Sullivan III wrote: On Mon, 2009-08-03 at 12:40 -0500, Tilghman Lesher wrote: On Monday 03 August 2009 12:30:12 pm John A. Sullivan III wrote: On Mon, 2009-08-03 at 13:04 -0400, John A. Sullivan III wrote: Hello, all. After reading the README, UPGRADE.txt, and a quick tour through google, is it safe to assume to upgrade from 1.6.1.1 to 1.6.1.2, one simply compiles and installs over the old installation being careful to NOT install the sample files? Thanks - John We've hit a problem even before installing. We're using Zimbra as IMAP storage for our voicemails. When we run make menuselect in 1.6.1.2, the IMAP storage option is disabled (XXX). When we run menuselect in 1.6.1.1 on the same system, it is available and enabled. Did we miss something or is this a bug? Thanks - John Re-run configure on 1.6.1.1. It's likely that the option will go away, as the dependency is no longer met. I'm not sure I understand. Nothing has changed to make the dependency fail. This is the same device where we are quite successfully running voicemail in IMAP - John Very strange, I did as you suggested and sure enough, IMAP is disabled as an option in 1.6.1.1. I'll have to dig a little deeper as I believe the most we may have done was a yum update! Thanks - John Ah, I remember now and shame on us for not documenting it. I'll record it here in case someone else hits the same thing. The version of libc-client that ships with CentOS 5.3 is too old to work with Zimbra. We thus needed to use the later imap-2007e version. configure needs to point to it, in our case: ./configure --with-imap=/home/compuser/Asterisk/imap-2007e I'll sheepishly add that to our internal documentation now :-( Would anyone mind answering the original question, though. Is it correct to simply compile and install over 1.6.1.1 to upgrade to 1.6.1.2? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] dahdi_dummy soft lockup in dahdi-linux-2.2.0.2
Hello, all. We attempted an upgraded from 1.6.1.1 to 1.6.1.2 today including upgrading dahdi-linux from 2.1.0.4 to 2.2.0.2 and dahdi-tools from 2.1.0.2 to 2.2.0. After rebooting, we receive: Aug 3 17:20:44 pbx01 kernel: BUG: soft lockup - CPU#2 stuck for 10s! [swapper:0] Aug 3 17:20:44 pbx01 kernel: CPU 2: Aug 3 17:20:44 pbx01 kernel: Modules linked in: ipv6 xfrm_nalgo crypto_api autofs4 dahdi_dummy(U) xpp_usb(U) xpp(U) wctc4xxp(U) dahdi_transcode(U) wcb4xx Aug 3 17:20:44 pbx01 kernel: Pid: 0, comm: swapper Tainted: G 2.6.18-128.2.1.el5 #1 Aug 3 17:20:44 pbx01 kernel: RIP: 0010:[80064c08] [80064c08] _spin_unlock_irqrestore+0x8/0x9 Aug 3 17:20:44 pbx01 kernel: RSP: 0018:81011fc87e98 EFLAGS: 0246 Aug 3 17:20:44 pbx01 kernel: RAX: RBX: RCX: Aug 3 17:20:44 pbx01 kernel: RDX: 8020 RSI: 0246 RDI: 883ec3b0 Aug 3 17:20:44 pbx01 kernel: RBP: 81011fc87e10 R08: 8857f420 R09: 810080bf8b00 Aug 3 17:20:44 pbx01 kernel: R10: R11: 0282 R12: 8005dc8e Aug 3 17:20:44 pbx01 kernel: R13: 0002 R14: 80077533 R15: 81011fc87e10 Aug 3 17:20:44 pbx01 kernel: FS: 41e57940() GS:81011fc5ce40() knlGS: Aug 3 17:20:44 pbx01 kernel: CS: 0010 DS: 0018 ES: 0018 CR0: 8005003b Aug 3 17:20:44 pbx01 kernel: CR2: 2b75e2c8c0a0 CR3: 00201000 CR4: 06e0 Aug 3 17:20:44 pbx01 kernel: Aug 3 17:20:44 pbx01 kernel: Call Trace: Aug 3 17:20:44 pbx01 kernel: IRQ [883e117b] :dahdi:dahdi_receive+0x7ce/0x7f9 Aug 3 17:20:44 pbx01 kernel: [885762ee] :dahdi_dummy:dahdi_dummy_timer+0xb3/0xfd Aug 3 17:20:44 pbx01 kernel: [8857623b] :dahdi_dummy:dahdi_dummy_timer+0x0/0xfd Aug 3 17:20:44 pbx01 kernel: [80094e14] run_timer_softirq+0x133/0x1af Aug 3 17:20:44 pbx01 kernel: [80011fc3] __do_softirq+0x89/0x133 Aug 3 17:20:44 pbx01 kernel: [8005e2fc] call_softirq+0x1c/0x28 Aug 3 17:20:44 pbx01 kernel: [8006cada] do_softirq+0x2c/0x85 Aug 3 17:20:44 pbx01 kernel: [8006b287] default_idle+0x0/0x50 Aug 3 17:20:44 pbx01 kernel: [8005dc8e] apic_timer_interrupt+0x66/0x6c Aug 3 17:20:44 pbx01 kernel: EOI [8006b2b0] default_idle+0x29/0x50 Aug 3 17:20:44 pbx01 kernel: [80048d9e] cpu_idle+0x95/0xb8 Aug 3 17:20:44 pbx01 kernel: [80076c3f] start_secondary+0x45a/0x469 Aug 3 17:20:44 pbx01 kernel: We are running on fully patched CentOS 5.3: [r...@pbx01 Asterisk]# uname -a Linux pbx.mycompany.com 2.6.18-128.2.1.el5 #1 SMP Tue Jul 14 06:36:37 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux We are assuming this is a dahdi-linux problem. Our procedure was a straightforward make (as unprivileged user) and make install (as root). Reverting to the previous versions on the same kernel resolves the problem - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Sound through NAT issue
On Thu, 2009-07-30 at 16:19 +0100, Paulo Santos wrote: Hello everyone, I'm having a hard time configuring my router to forward asterisk traffic correctly. I have the following ports being forwarded to asterisk: 5060, 1-2 Now, I can register the accounts when outside the network and I can call every extension that is inside the network. The problem is that I can't ear anything nor can the phones inside the network phone the outside phone. Is there any port I'm forgetting to forward? snip What happens if you set canreinvite=no in sip.conf or the appropriate sip configuration file? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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
[asterisk-users] Call history problems from B2BUA
Hello, all. Alas, another convoluted question. All the simple things are, well, simple so I suppose we only need to trouble the list with squirrely problems! We've noticed a call history problem when using Asterisk where the call history on the Snom phones (with which we are very pleased) reflects the number of the PBX extension used by the B2BUA to dial the end point. I assume the same would be true of any B2BUA. There are two flavors of this problem. In one version, addresses outside of the PBX work fine but calls from different contexts within the same multi-tenant PBX cannot be returned from call history. In the other, all call history is broken. I know that sounds confusing so let me illustrate. Here is the first scenario. Imagine a multi-tenant PBX with two tenants - mycompany.com and yourcompany.com. Each allows direct inbound SIP dialing to addresses such as us...@mycompany.com. The tenants live in separate contexts within the PBX and cannot see each other's contexts for both security and because they have some overlapping extensions. Internally, us...@mycompany.com uses extension 312 and thus is accessed at 3...@pbx.mycompany.com. His Snom phone is us...@10.2.2.20. us...@yourcompany.com uses extension 15 and is this accessed at 1...@pbx.yourcompany.com while his Snom phone is at 10.1.1.10. pbx.mycompany.com and pbx.yourcompany.com map to the same IP address - the multi-tenant PBX. us...@mycompany.com makes a direct SIP call to us...@yourcompany.com. us...@mycompany.com connects to Asterisk as its outbound proxy and is associated with 3...@pbx.mycompany.com. 3...@pbx.mycompany.com then calls us...@10.1.1.10 (IP address of the Snom phone) and we have a successful B2BUA call. us...@yourcompany.com wants to call back us...@mycompany.com so they go to their call history. The call shows up as from user1 3...@pbx.mycompany.com (actually, it seems to use the resolved IP address). user2 highlights the call in the history, presses check and the call fails because there is no 312 in any accessible context. How would we get the call history to show us...@mycompany.com instead of 3...@x.x.x.x? This leads to the second scenario which is actually our preferred configuration. We set fromuser and fromdomain in sip.conf. This is preferable to use because we use different IDs internally and externally so we never expose internal IDs to the world. Thus, user1's sip ID might be 1user but his public sip address is us...@mycompany.com. Thus, we would like to overwrite his outbound identity to be us...@mycompany.com. That's why we originally set fromuser and fromdomain but this backfired. Now user2 sees user1's call as coming from user1 us...@yourcompany.com because user2's sip configuration says to set fromuser=user2 and fromdomain=yourdomain.com. It seems when the B2BUA sets up the leg to user2, it thinks it is coming from user2's extension. In fact, this even breaks outside numbers. If I dial in from the PSTN, e.g., 207-111- in the US, the call history shows: 207111 sip:us...@yourcompany.com If user2 attempts to dial from their call history, they will dial themselves! How do we get this to work properly? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] outbound calls not reaching vitelity
On Tue, 2009-07-28 at 13:33 -0500, Tom Poe wrote: Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom snip I'm not using pbxinaflash but I am using Vitelity and have had no problems at all - in fact very happy with them. They should have given you a management portal for your account probably portal.vitelity.net. In there, there is an option to open a trouble ticket. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] TLS Manager
Thank you although that seems a bit strange. Does one simply concatenate them together or is it really looking for a PKCS#12 file? Thanks - John On Sun, 2009-07-26 at 10:03 -0700, Eric Chamberlain wrote: The pem file should contain both the private key and the certificate. On Jul 24, 2009, at 4:08 PM, John A. Sullivan III wrote: Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path to the certificate. ; sslcipher=cipher string It errors as I expect it would: pbx*CLI manager reload == Parsing '/etc/asterisk/manager.conf': == Found SSL cert error /etc/pki/tls/certs/pbxc.pem How does one specify the private key for the manager.conf file? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Cell phones and (no) rings
Hello, all. Our first major asterisk system is just about ready for production. However, we noticed that our outbound SIP callers did not receive rings when dialing cell phones. Land lines were fine. We fixed this by setting progressinband=no in sip.conf. However, I gather this places extra load on the asterisk server and I'm not sure that it always conveys accurate status (e.g., busy, non-North-American destination). Is there a better way to ensure our dialers hear a ring when dialing cell numbers? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] TLS Manager
After some testing and false starts, it looks like PKCS#12 does not work but simple concatenation does. Thanks - John On Mon, 2009-07-27 at 06:38 -0400, John A. Sullivan III wrote: Thank you although that seems a bit strange. Does one simply concatenate them together or is it really looking for a PKCS#12 file? Thanks - John On Sun, 2009-07-26 at 10:03 -0700, Eric Chamberlain wrote: The pem file should contain both the private key and the certificate. On Jul 24, 2009, at 4:08 PM, John A. Sullivan III wrote: Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path to the certificate. ; sslcipher=cipher string It errors as I expect it would: pbx*CLI manager reload == Parsing '/etc/asterisk/manager.conf': == Found SSL cert error /etc/pki/tls/certs/pbxc.pem How does one specify the private key for the manager.conf file? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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 ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Inquiry abount Asterisk extensions.conf
http://www.voip-info.org/wiki/view/Asterisk+Extension+Matching http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns http://wiki.snom.com/Settings/overlap_dialing Hope this helps - John On Sun, 2009-07-26 at 05:07 +0100, hadi motamedi wrote: Dear Leif Can you please provide us with more details on this Overlap Dialing phillosophy ? Regards H.Motamedi On Wed, Jul 22, 2009 at 1:15 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote: John Novack wrote: Can you please let us know how we can modify our Asterisk extensions.conf file so it interprets the subscriber dialed digits in one-by-one digit manner . At its current configuration , it interprets them in an whole packet . I mean , say the subscriber dials as 665 so we need Asterisk to send it to the peer switch as 6,6,5,0,0,0,0 but not as one 665 packet . Curious - Why? What is the peer switch and why does it have this requirement? That's a funny way of answering the question :) I *think* what he wants is overlap dialing. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ -- 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 ___ -- 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 -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] TLS Manager
Hello, all. After many pages of googling and testing in the lab, I'm still a bit perplexed about how to implement tls protection for the asterisk manager. manager.conf allows one to specify the cert file but one normally must also specify the private key file. If I simply enter the cert file: sslenable=yes sslbindport=5038 sslbindaddr=172.x.x.8 sslcert=/etc/pki/tls/certs/pbxc.pem ; path to the certificate. ; sslcipher=cipher string It errors as I expect it would: pbx*CLI manager reload == Parsing '/etc/asterisk/manager.conf': == Found SSL cert error /etc/pki/tls/certs/pbxc.pem How does one specify the private key for the manager.conf file? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Audio lost on reinvite
Hello, all. We are having a problem where audio for sip channels is dropping upon reinvite. Perhaps it reflects a misunderstanding of what reinvite does. We are running Asterisk 1.6.1.1 on CentOS 5.3. SIP is set to canreinvite=nonat. We have tried RTP with strictrtp set to both yes and no. We have also tried extending the Asterisk rtp port range to accommodate the differing default ranges of the soft phones (Twinkle on Linux, 3CX on Windows). Testing revealed no problems when the soft phones we used for testing were on the same physical and logical network. Once we moved the soft phones to OpenVPN connections (same logical network but different physical media), the call is setup, the receiver hears the caller for the briefest instant (we are assuming the first reinvite), the caller hears the receiver for some time (perhaps 20 - 30 seconds) and then the receiver's voice disappears, too. At that very moment, there is another redirect and RTP traffic starts on a different set of ports from the receiver. Packet traces revealed RTP packets flowing from the receiver to Asterisk but no packets coming back from Asterisk except ICMP service unreachable for port 8000 (the new port after the second reinvite). It's as if Asterisk does not recognize the ports after the reinvite. We were actually surprised to see the packets flowing between the soft phones and Asterisk as I would have thought the reinvite would direct traffic to flow directly between the soft phones - both of whom can ping each other and are on the same physical network. We've traced from the perspective of the end points, the gateway, and Asterisk. All show the same pattern: the caller is having a dialog with Asterisk whereas the receiver is having a monolog - no packets back from Asterisk. Could someone explain why were are losing the audio, why we see a dialog on one side but a monolog on the other, and why we are not seeing the reinvite redirect the packet stream to be directly between the soft phones. Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Connecting multiple office with multiple servers
On Tue, 2009-07-21 at 09:46 -0400, Ekelund, Bryan wrote: Greetings all, I currently manage a two-server asterisk system that connects two of our offices. Running 1.4 on CentOS 5.2 on both sides. We use Polycom 501 phone and register the phones to both systems, and use SIP peering to interconnect the two systems. We had been using IAX, but found that for some reason we were having trouble keeping that data stream in out QOS. Sometime in the near future, we are planning on integrating two of our other remote offices, each with their own asterisk server, into this network. I would like to have the phones register to two servers, but be able to be seen by all four. I have been experimenting with OpenSips/Kamailio as a registration server and forwarding all SIP requests to the appropriate office, but that may have a larger learning curve than I would like for the timeframe I am working with. I am looking at DUNDi and am thinking that this might be the way to merge these systems together and share the registrations between the servers. I am sure someone has experience with this type of setup, and I was hoping that I could confirm that DUNDi might be the way to go, or if not, maybe point me in the right direction. snip I wonder if one could use a realtime setup and store the registrations in a common database. I believe I read that is how one shares them between Kamailio and Asterisk - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] CyberData SIP-enabled VoIP Intercom
On Mon, 2009-07-20 at 00:35 +0200, FiNKu wrote: Hi, Did anyone have any experience with CyberData SIP-enabled VoIP Intercom units please? Are they any good? Can you recommend anything better? snip We configured our very first one two weeks ago and are still awaiting installation. If I recall correctly, the only quirky bits were we had to crank down the registration time to something like three minutes, set qualify=no, and canreinvite=no. Other than that, it seemed pretty straightforward but I can't yet report on how well it worked - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] dialplan number matching
On Fri, 2009-07-17 at 02:11 -0700, Vieri wrote:
Hi,
How can I match an extension ending with 3 (just an example but applicable
to any other digit, including * or #)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard dialplan (end with: $)?
Thanks,
Vieri
snip
I haven't tried it but I wonder if one could use a regex pattern match
in a GotoIf statement and then pass the result to another context using
${EXTEN}? Just a thought - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
___
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[asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
The second was fixed by backporting a patch from SVN but we still have
the first problem.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE, 0.5)
in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'
We then see the park timeout and fail to return
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Oops! Thought I had changed to address! My apologies - John
On Fri, 2009-07-17 at 13:29 -0400, John A. Sullivan III wrote:
Hello, all. My apologies for troubling the developer list as an end
user but we were not able to resolve this issue on the user list and it
is smelling like a possible bug when using multi-tenant call parking.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
The second was fixed by backporting a patch from SVN but we still have
the first problem.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE,
0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new
stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new
stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye)
in new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn
Re: [asterisk-users] dialplan number matching
On Fri, 2009-07-17 at 12:56 -0700, Vieri wrote:
--- On Fri, 7/17/09, John A. Sullivan III jsulli...@opensourcedevel.com
wrote:
Hi,
How can I match an extension ending with 3 (just an
example but applicable to any other digit, including * or
#)?
exten = _ZX.3,n,...
exten = _ZX.#,n,...
(the above does not work)
Can regular expressions be used in the standard
dialplan (end with: $)?
Thanks,
Vieri
snip
I haven't tried it but I wonder if one could use a regex
pattern match
in a GotoIf statement and then pass the result to another
context using
${EXTEN}? Just a thought - John
Thanks, I'll think about it but I don't think it will apply efficiently to
the goal I describe here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg227054.html
Anyway, I solved my early-dial issue by creating a special context where
I Read() the user's input until he/she presses #. It's not as elegant as
having Asterisk match regular expressions or do something like exten =
_00ZX.#,n,... but I'll settle with it.
snip
I am very new to Asterisk so you probably know far more than I and I
have never used the regex logic but what about something like:
exten = _00ZX.,n,GotoIf($[${EXTEN}:.*3$]?:no3)
exten = _00ZX.,n,DO SOMETHING
exten = _00ZX.,n(no3),NoOp()
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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Re: [asterisk-users] Voicemail login incorrect
On Thu, 2009-07-16 at 10:57 -0400, Zaheer Master wrote: Hi all, I'm having trouble with voicemail on my *NOW 1.5/FreePBX box. I have enabled voicemail in the extensions area, and set the default password. However, every time I try to log in with a mailbox and password, I get the message login incorrect. I've tried changing the voicemail password, and also disabling and re-enabling the voicemail feature. What else can I do to set up the voicemail? Also, I've left the VM Context as default and the mailbox is 101. snip If you set your Asterisk console to a verbose mode, what password do you see passed to the voicemail application? We recently noticed 3CX softphone users with multiple options set for DTMF were sending duplicate DTMF signals to our voicemail resulting in the same problems you are seeing, e.g., 1234 would be sent as 11223344. Just a guess - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] QoS
On Wed, 2009-07-15 at 08:10 -0500, Danny Nicholas wrote: In my shop, we got a better router to support QOS and configured our Polycom phones to always request highest levels (UDP gets 6, everything else gets 3). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, July 14, 2009 5:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] QoS Howdy, Getting ready to play with QoS settings. We have an asterisk 1.4.23 server running in a colo bunker in the US Virgin Islands under a large radio tower. That tower has multiple sector radio/antenna pairs that blanket a valley in 802.11a. The customers have directed dishes aimed at the sector antennas, mounted on their roofs. This setup has been working great for their broadband access for many years. Now we want to sell voice services on top of this infrastructure, and it works fine too, until they start some data intensive process on the customer end, like bittorrent :) We would like to avoid these problems by properly setting up packet prioritization between the customer and the sector radios, which we have control over. Any links to share to get us started? Basically from zero? :) snip I'm entirely unfamiliar with your environment and very new to Asterisk so please take what I say with a large dose of skepticism. We elected to move CoS right to the core switch and tried to keep it consistent throughout the path. Our environment is still pretty simple so we are using HP Procurve 2810 switches. Asterisk sits on its own VLAN. We believe we found some conflict between typical DSCP settings and Linux routers / firewalls in their default state. We initially set our systems to use Expedited Forwarding for both SIP and audio RTP. I believe this is b8 in Asterisk and 184 for our Snom phones. This sets the bits in the DSCP field as 101110. However, the default Linux packet prioritization (pfifo_fast) is looking at only the last three bits of that field (because it is not actually using DSCP but the ToS bits). It sees 110 and that middle 1 causes it to place the packets in band1 which is the default processing rather than band0 which is priority processing. We thus changed the DSCP header to 101100 (b0 in Asterisk and 176 in Snom). We believe this will cause default Linux routers / firewalls using pfifo_fast to process these packets in band0 (high priority). We then returned to our switch and told it to map DSCP header 101100 to its highest priority path. Thus, we should now have consistent CoS from the servers to the switches to the firewalls to the phones. Since our connection to Asterisk is via VPN, we also ensure the ToS bits are passed to the VPN header (be it IPSec or OpenVPN). I know that sounds dreadfully complicated but that is how we did it. If someone sees a better way or if we are unnecessarily complicating it, please let us know. If you need more information, I can probably post some of our internal documentation. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] how to enable dial to a...@asterisk.blurb.com
On Wed, 2009-07-15 at 14:34 +1000, Alex Samad wrote:
Hi
The subject line says it all how do I enable this style of call.
Pointers to the dns setup and asterisk setup would be great
or even search words for google, as I am not sure how to search for this
type of request.
Alex
snip
If I understand what you are seeking, you can try these URIs:
http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
http://www.blyon.com/blog/index.php/2009/06/22/p2p-sip-uri-dialing/
However, I found I changed mine substantially. I am very new to
Asterisk so if this seems like a silly idea, it probably is and I would
appreciate being told so! We generally use numeric extensions - old
habits I suppose. We found that the catch-all _. for uri dialing was
also catching mis-dialed extensions. That led us to this solution:
[dial-uri] ; Always include this last because of its broad matches
exten = _[a-zA-Z0-9].,1,GotoIf($[${SIPDOMAIN}!=pbx01.ssiservices.biz]?:_.,1)
; non-URIs will automatically append @pbx01.ssiservices.biz
; this logic separates mistyped extensions from valid URI attempts
exten = _[a-zA-Z0-9].,n,Macro(uridial,${ext...@${sipdomain})
exten = _.,1,Answer(0.5)
exten = _.,n,Playback(im-sorry)
exten = _.,n,Wait(0.0.5)
exten = _.,n,Playback(you-dialed-wrong-number)
exten = _.,n,Wait(0.4)
exten = _.,n,Playback(vm-goodbye)
exten = _.,n,Hangup()
Here is the macro:
[macro-uridial]
exten = s,1,NoOp(Calling remote SIP peer ${ARG1})
exten = s,n,Dial(SIP/${ARG1},60)
exten = s,n,Congestion()
As I think about it, I wonder if that NoOp should be replace with a
Verbose. In any event, I hope this helps.
Oh, of course, this is for outbound. For inbound, one creates explicit
entries for each SIP URI and map these to the appropriate extensions.
For example, for users, we typically map to their email address (which
is different than their internal ID; for security purposes, publicly
exposed IDs are different from internally used IDs). We also create
direct SIP extensions for things like voicemail, office numbers, world
headquarters, so that direct SIP calls can be used just like regular
calls and enter our calling tree:
[a100in] ; direct inbound SIP dialing
exten = conference,1,Goto(a100pub,6000,1)
exten = someone,1,Goto(a100pub,314,1)
exten = helpdesk,1,Goto(a100pub,302,1)
exten = someoneelse,1,Goto(a100pub,312,1)
exten = mycompany-hq,1,Goto(a100pub,9,welcome)
exten = mycompany-europe,1,Goto(a100pub,9,welcome)
exten = mycompany-us,1,Goto(a100pub,9,welcome)
exten = vmail,1,Goto(a100pub,7000,1)
Since we are a secure, multi-tenant environment, we do not place these
in the default inbound context for sip. Instead, we only allow
designated domains in our sip.conf and specify a separate inbound
context for each which lands them into these sip directories, e.g., :
autodomain=no
domain=pbx01.mycompany.com
domain=172.x.y.8
; define client domains
domain=yourcompany.com,a100in
domain=theircompany.com,a10in
domain=pbx01.theircompany.com
allowexternaldomains=yes
Hope this helps. If someone sees a better way, please say so. Thanks -
John
--
John A. Sullivan III
Open Source Development Corporation
Street Preacher: Are you SAVED?!!
Educated Skeptic: Saved from WHAT?!!
Educated Believer: From our selfishness that hurts the ones we love
and condemns us to an eternity of hurting each other.
http://www.spiritualoutreach.com
Christianity that makes sense
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Re: [asterisk-users] Caller ID (name) - where does it come from?
On Tue, 2009-07-07 at 16:54 -0400, Barry D. Hassler wrote: This is all excellent information. My primary issue is for calls that are placed FROM my client's PBX, via VOIP provider (Teliax). The recipients of those calls are the ones that are not getting the proper CNAM information as the call comes in. We just recently ported the client's POTS lines to VOIP, and with the exception of this issue, all is working well. But, my client is really unhappy that their callerID NAME isn't showing up. snip I was very curious about this myself. We successfully set the CallerID number by creating different contexts for our various offices and using a Set(CALLERID(num)=x) call. But we could not set the name so I asked our new carrier (Vitelity - with whom we have been quite pleased thus far). This is their response to us: We can have the name set for this number, however there is a one time passthrough charge of $xx per number for the update. Outbound caller ID is updated into a national database called LIDB (line information database), it is the final terminating provider that is responsible for querying this database and delivering it to their customers. -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our
brand new
multi-tenant setup seems to be working smoothly from start
to finish
with just a bump or two. The biggest is parking. Now that
we got most
kinks worked out, I'm a little more comfortable in trying to
resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default
group no matter
what we do.
I haven't tested this.
2. When the parked call timer expires, the callback to
the original
callee fails because a | delimiter is used in the
Dial()
function.
This has been fixed in the 1.6.1 SVN, and you will have to back port a
patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
snip
Hello, all. I applied the patch as graciously supplied by Jonathan. It
solves the callback problem of the | delimited Dial parameters but the
basic problem of pulling parking places from the default parking lot
still exists. Same results as last time:
Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
What are we doing wrong? Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
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[asterisk-users] Zimbra IMAP authentication - SOLVED
Hello, everyone. No need to read this message. I'm posting for
documentation for other poor, ignorant slobs like me who are struggling
to pull together the many technologies to make converged networks
happen. Hopefully, this will help save someone else the time I spent.
I started the below email until I realized I had solved multiple parts
of a compound problem but not all at the same time. When I put them
together in the right order, it worked.
I did not understand that I needed to use AUTHENTICATE PLAIN and that
such authentication is a single string which pertains to the user and
not the authuser. Then, once I got the right password parameter name
and figured out that I could not use a distribution list rather than a
real email account for shared voice mail (duh), it all fell into place.
The configuration in the examples below works. It's a wonder to behold.
Thanks Asterisk developers - John
Hello, all. I'm having a nasty time trying to integrate Asterisk and
Zimbra for voice mail. No matter whether I use imappassword=,
imappasswd=, or imapsecret=, I get these errors:
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
Retrying PLAIN authentication after AUTHENTICATE failed
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
Retrying PLAIN authentication after AUTHENTICATE failed
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 2 19:54:26] ERROR[26609]: app_voicemail.c:2309 mm_log: IMAP Error: Can
not authenticate to IMAP server: AUTHENTICATE failed
[Jul 2 19:54:26] ERROR[26609]: app_voicemail.c:1669 messagecount: Houston we
have a problem - IMAP mailstream is NULL
My voicemail.conf file has lines such as:
10 = x,Some
User,,,imapuser=per...@somewhere.com|imappassword=Y2xlcmljYWxAZWJjLWNvLmNvbXgAemltYnJhbWFuAFNTIVMzcnZpY2VzcEBzc3BocmFzZQ==
I can authenticate via telnet with . authenticate plain using these
passwords. If it's of any help to anyone, I put together a small script
to produce them:
#!/bin/bash
# Copyright 2009 by John A. Sullivan III, SSI Services, LP
# This script takes a file with a list of email accounts (accountfile) and
# produces a file containing Zimbra PLAIN AUTHENTICATION passwords
# (accountfile.pauth) in the current directory.
# Thus, be sure you have read rights where you run this script.
if [ -z ${1} ];then
echo usage: $(basename ${0}) accounts file name
exit 5
fi
read -p What is the admin email account name? ADMIN
echo Thank you
read -s -p Now what is the admin's password? APW
LINE=
OFILE=$(basename ${1}).pauth
: ${OFILE}
while read EADD
do
echo ${EADD}
LINE=$(printf ${EADD}\000${ADMIN}\000${APW} | openssl base64 | tr -d
'\n')
echo -e ${EADD}\t${LINE}\n ${OFILE}
done ${1}
Here is a portion of voicemail.conf:
pollmailboxes=yes
pollfreq=60
; IMAP voice mail storage
imapserver=zimbra.ssiservices.biz
imapport=7143 ; Using the Zimbra IMAP proxy at 143 on this station - real IMAP
listens on 7143
expungeonhangup=yes
imapfolder=INBOX
imapflags=notls
;authuser=mana...@ssiservices.biz
;authpassword=password
imapgreetings=no
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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[asterisk-users] Converged mail box sizes
Just a thought as we explore the brave new world of converged voice and emails. Voice mail boxes typically hold a very small number of messages while email folders contain thousands. Do we need to rethink the traditionally small limits on voice mail boxes when storing in IMAP or are the messages counted separately? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Zimbra IMAP authentication - SOLVED
On Thu, 2009-07-02 at 20:59 -0400, John A. Sullivan III wrote:
Hello, everyone. No need to read this message. I'm posting for
documentation for other poor, ignorant slobs like me who are struggling
to pull together the many technologies to make converged networks
happen. Hopefully, this will help save someone else the time I spent.
I started the below email until I realized I had solved multiple parts
of a compound problem but not all at the same time. When I put them
together in the right order, it worked.
I did not understand that I needed to use AUTHENTICATE PLAIN and that
such authentication is a single string which pertains to the user and
not the authuser. Then, once I got the right password parameter name
and figured out that I could not use a distribution list rather than a
real email account for shared voice mail (duh), it all fell into place.
The configuration in the examples below works. It's a wonder to behold.
Thanks Asterisk developers - John
Hello, all. I'm having a nasty time trying to integrate Asterisk and
Zimbra for voice mail. No matter whether I use imappassword=,
imappasswd=, or imapsecret=, I get these errors:
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
Retrying PLAIN authentication after AUTHENTICATE failed
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
Retrying PLAIN authentication after AUTHENTICATE failed
[Jul 2 19:54:26] WARNING[26609]: app_voicemail.c:2306 mm_log: IMAP Warning:
SECURITY PROBLEM: insecure server advertised AUTH=PLAIN
[Jul 2 19:54:26] ERROR[26609]: app_voicemail.c:2309 mm_log: IMAP Error: Can
not authenticate to IMAP server: AUTHENTICATE failed
[Jul 2 19:54:26] ERROR[26609]: app_voicemail.c:1669 messagecount: Houston we
have a problem - IMAP mailstream is NULL
My voicemail.conf file has lines such as:
10 = x,Some
User,,,imapuser=per...@somewhere.com|imappassword=Y2xlcmljYWxAZWJjLWNvLmNvbXgAemltYnJhbWFuAFNTIVMzcnZpY2VzcEBzc3BocmFzZQ==
I can authenticate via telnet with . authenticate plain using these
passwords. If it's of any help to anyone, I put together a small script
to produce them:
#!/bin/bash
# Copyright 2009 by John A. Sullivan III, SSI Services, LP
# This script takes a file with a list of email accounts (accountfile) and
# produces a file containing Zimbra PLAIN AUTHENTICATION passwords
# (accountfile.pauth) in the current directory.
# Thus, be sure you have read rights where you run this script.
if [ -z ${1} ];then
echo usage: $(basename ${0}) accounts file name
exit 5
fi
read -p What is the admin email account name? ADMIN
echo Thank you
read -s -p Now what is the admin's password? APW
LINE=
OFILE=$(basename ${1}).pauth
: ${OFILE}
while read EADD
do
echo ${EADD}
LINE=$(printf ${EADD}\000${ADMIN}\000${APW} | openssl base64 | tr -d
'\n')
echo -e ${EADD}\t${LINE}\n ${OFILE}
done ${1}
Here is a portion of voicemail.conf:
pollmailboxes=yes
pollfreq=60
; IMAP voice mail storage
imapserver=zimbra.ssiservices.biz
imapport=7143 ; Using the Zimbra IMAP proxy at 143 on this station - real
IMAP listens on 7143
expungeonhangup=yes
imapfolder=INBOX
imapflags=notls
;authuser=mana...@ssiservices.biz
;authpassword=password
imapgreetings=no
Hmm . . . I shouldn't have celebrated so quickly. It suddenly all came
crashing down and I don't understand why. When I do a packet trace, the
strings being passed as the AUTHENTICATE PLAIN tokens are nothing like
the strings in the voicemail.conf file! Does the conf file want them in
a different format or is it doing something else with them?
Ah, it looks like another part of a compound problem - the age of the
c-client library. I am running on CentOS 5.3 but the library it uses is
from 2004. Perhaps it is the combination of very old libc-client and
very new Zimbra. I installed the latest recommended versions of
c-client (2007e), recompiled, went back to using a single authuser and
authpassword and all is working!
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
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Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Thu, 2009-07-02 at 17:42 -0400, John A. Sullivan III wrote:
On Wed, 2009-07-01 at 07:17 -0700, Jonathan Thurman wrote:
On Tue, Jun 30, 2009 at 11:53 PM, John A. Sullivan III
jsulli...@opensourcedevel.com wrote:
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our
brand new
multi-tenant setup seems to be working smoothly from start
to finish
with just a bump or two. The biggest is parking. Now that
we got most
kinks worked out, I'm a little more comfortable in trying to
resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default
group no matter
what we do.
I haven't tested this.
2. When the parked call timer expires, the callback to
the original
callee fails because a | delimiter is used in the
Dial()
function.
This has been fixed in the 1.6.1 SVN, and you will have to back port a
patch until these changes are rolled into another release. I was
disappointed that more bug fixes were not included in 1.6.1.1.
snip
Hello, all. I applied the patch as graciously supplied by Jonathan. It
solves the callback problem of the | delimited Dial parameters but the
basic problem of pulling parking places from the default parking lot
still exists. Same results as last time:
Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
What are we doing wrong? Thanks - John
By the way, I did try it both ways - creating the lot from features.conf
using 700 and creating my own 700 extension for parking using CHANNEL.
Neither worked. Thanks - John
--
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com
http://www.spiritualoutreach.com
Making Christianity intelligible to secular society
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[asterisk-users] Multi-tenant parking broken in 1.6.1.1?
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE, 0.5)
in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on 'SIP/jasiii-cc05ceb8'
We then see the park timeout and fail to return to the original
Re: [asterisk-users] Multi-tenant parking broken in 1.6.1.1?
On Wed, 2009-07-01 at 02:17 -0400, John A. Sullivan III wrote:
Hello, all. With the assistance of very helpful folks, our brand new
multi-tenant setup seems to be working smoothly from start to finish
with just a bump or two. The biggest is parking. Now that we got most
kinks worked out, I'm a little more comfortable in trying to resolve
this.
There seem to be two problems:
1. Parking assigns parking spaces from the default group no matter
what we do.
2. When the parked call timer expires, the callback to the original
callee fails because a | delimiter is used in the Dial()
function.
Perhaps we have configured it incorrectly. Here is the pertinent
section from features.conf:
[parkinglot_a10] ; EBC
context = a10parking
parkpos = 101-110
;parkext = 100
findslot = next
[parkinglot_a100] ; SSI
context = a100parking
;parkext = 1000
parkpos = 1001-1020
findslot = next
If I understand this correctly, the parkinglog_a100 would be the channel
variable and a100parking the context into which parking extensions are
placed.
We set the channel parameter in sip.conf:
[a100](!,common)
context=a100
vmext=999
parkinglot=parkinglot_a100
subscribecontext=a100
accountcode=a-0100
fromdomain=ssiservices.biz
[userx](a100)
mailbox=...@a100,x...@a100
secret=something
callerid=John A. Sullivan III xxx
fromuser=userid
and we included the context in extensions.conf:
[a100] ; SSI
exten = 911,1,Macro(emergency-US,xx)
exten = 9911,1,Macro(emergency-US,xx)
exten = ,1,VoiceMailMain(${CALLERID(num)}...@a100) ; Direct mail
retrieval
include = a100pub
include = a100conf
include = a100parking
include = US-international
include = dial-uri
We also tried Set(CHANNEL(parkinglot)=parkinglot_a100). We also tried
creating our own parking which yielded interesting data but not
solution.
Here is the console output using the regular setup described:
Call comes in and is answered:
-- SIP/gss-cc01c918 answered SIP/localhost-cc002cf8
-- Native bridging SIP/localhost-cc002cf8 and SIP/gss-cc01c918
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Call is parked:
-- Executing [...@a100:1] Park(SIP/gss-cc05ceb8, ) in new stack
== Parked SIP/gss-cc05ceb8 on 701 (lot default). Will timeout back to
extension [a100] s, 1 in 60 seconds
-- Added extension '701' priority 1 to parkedcalls (0x2cca3f70)
-- SIP/gss-cc05ceb8 Playing 'digits/7.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/0.ulaw' (language 'en')
-- SIP/gss-cc05ceb8 Playing 'digits/1.ulaw' (language 'en')
-- Started music on hold, class 'default', on SIP/gss-cc05ceb8
I'm not sure what is happening here but I think this is the original
callee releasing the call. I don't know what the ZOMBIE extension is
about:
== Spawn extension (a100, s, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Auto fallthrough, channel 'Parked/SIP/gss-cc05ceb8ZOMBIE' status is
'UNKNOWN'
-- Executing [...@a100:1] Answer(Parked/SIP/gss-cc05ceb8ZOMBIE,
0.5) in new stack
== Spawn extension (a100, h, 1) exited non-zero on
'Parked/SIP/gss-cc05ceb8ZOMBIE'
-- Stopped music on hold on SIP/gss-cc05ceb8
-- Stopped music on hold on SIP/localhost-cc002cf8
-- Started music on hold, class 'default', on SIP/localhost-cc002cf8
== Spawn extension (macro-common, s, 1) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE' in macro 'common'
== Spawn extension (a100pub, 314, 2) exited non-zero on
'SIP/gss-cc05ceb8ZOMBIE'
== Using SIP RTP TOS bits 176
== Using SIP RTP CoS mark 5
Then we see the destination callee attempting to pick up the call and is
the output of our routine to catch misdialed/unknown extensions:
-- Executing [...@a100:1] GotoIf(SIP/jasiii-cc05ceb8, 0?:_.,1) in new
stack
-- Goto (a100,_.,1)
-- Executing [...@a100:1] Answer(SIP/jasiii-cc05ceb8, 0.5) in new
stack
-- Executing [...@a100:2] Playback(SIP/jasiii-cc05ceb8, im-sorry) in
new stack
-- SIP/jasiii-cc05ceb8 Playing 'im-sorry.ulaw' (language 'en')
-- Executing [...@a100:3] Wait(SIP/jasiii-cc05ceb8, 0.0.5) in new
stack
-- Executing [...@a100:4] Playback(SIP/jasiii-cc05ceb8,
you-dialed-wrong-number) in new stack
-- SIP/jasiii-cc05ceb8 Playing 'you-dialed-wrong-number.ulaw' (language
'en')
-- Executing [...@a100:5] Wait(SIP/jasiii-cc05ceb8, 0.4) in new stack
-- Executing [...@a100:6] Playback(SIP/jasiii-cc05ceb8, vm-goodbye)
in new stack
-- SIP/jasiii-cc05ceb8 Playing 'vm-goodbye.ulaw' (language 'en')
-- Executing [...@a100:7] Hangup(SIP/jasiii-cc05ceb8, ) in new stack
== Spawn extension (a100, _., 7) exited non-zero on 'SIP/jasiii-cc05ceb8'
-- Executing
