Re: [asterisk-users] SIP Hard Phone with SMS
2009/10/9 Juan E. Rodríguez jerdg...@gmail.com: Does any one know about a SIP hard phone capable of sending SMS messages (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? The Gigaset S675IP series of DECT/SIP phone has SMS capability, but not sure it can work with Asteris. /r ___ -- 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 Hard Phone with SMS
randulo schrieb: 2009/10/9 Juan E. Rodríguez jerdg...@gmail.com: Does any one know about a SIP hard phone capable of sending SMS messages (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? The Gigaset S675IP series of DECT/SIP phone has SMS capability, but not sure it can work with Asteris. Yes, they do. (app_sms) Make sure you have installed the latest FW. Before, they sent the SMS out on the analog port only. Hans ___ -- 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] Server-side scripting when SIP phones register
Hi, Using AMI, when a peer is set with Qualify=yes, it seems you can't make a difference between First-time registration and Re-registration. Looking at an AMI log, I saw: Re-registration (to be confirmed): Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/7266 PeerStatus: Registered Address: 10.10.20.109 Port: 5060 First-time registration (after pluging back a SIP phone): Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/7275 PeerStatus: Registered Address: 10.10.20.104 Port: 5060 If this 2nd message was different from the 1st one, it would possible to get this server-side feature. Example of first-time registration: Event: PeerStatus Privilege: system,all ChannelType: SIP Peer: SIP/7275 PeerStatus: Registered Address: 10.10.20.104 Port: 5060 SubPeerStatus: First I'm not enough aware of SIP channel internals to tell if it does make sense to hope to have this distinction made between registrations. 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
Re: [asterisk-users] Asterisk 1.6.1.6 suddently restarts ...
2009/10/8 Leif Madsen leif.mad...@asteriskdocs.org Please follow up on the issue tracker at http://issues.asterisk.org Thanks! Leif. I think I will. Two days ago we reverted back to 1.6.1.0 and Asterisk is running OK since. Hopefully, if this behaviour remains, we will give 1.6.1.7-rc2 a new shot and report gathered data to issue tracker. Thanks for helping ! Olivier wrote: As a follow up and unfortunately, I must say now that upgrading to 1.6.1.7-rc2 didn't help. We downgraded to a 1.6.1.0 with which we never met the problem we're facing 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 ___ -- 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] How to keep difference between 2 SIP-accounts/trunks from same server ??
What I have tried is : register = user1:pass...@server/yocan register = user2:pass...@server/itcenter extensions.conf : [default] exten = yocan,1,GoTo(user1,s,1) exten = itcenter,1,GoTo(user2,s,1) [user1] ... [user2] ... But the CLI shows : [Oct 9 09:28:52] -- Executing [...@macro-getiaxaccount:5] MYSQL(SIP/ITCENTER-3starsnet-076e4700, ... [Oct 9 09:28:52] -- Executing [...@macro-getiaxaccount:6] MacroExit(SIP/ITCENTER-3starsnet-076e4700,... [Oct 9 09:28:52] -- Executing [...@user1:9] NoOp(SIP/ITCENTER-3starsnet-076e4700, ... [Oct 9 09:28:52] -- Executing [...@user1:10] Dial(SIP/ITCENTER-3starsnet-076e4700, ... So the call comes into the right context... that's not the problem. But my CDR is messed up. The accountcode that I have set for user1 is always replaced for the accountcode I've set for user 2. [YOCAN-3starsnet] type=peer accountcode=user1_in [ITCENTER-3starsnet] type=peer accountcode=user2_in Is there yet another workaround ?! Is it not meant to host several SIP-accounts on 1 Asterisk-box that register to a SIP- provider ??? Jonas. On Thu, 2009-10-08 at 23:21 +0200, Dovid Bender wrote: - Original Message - From: jonas kellens To: Asterisk Mailing Sent: Thursday, October 08, 2009 15:20 Subject: [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ?? Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A -- Z. In my sip.conf : register = user1:pass...@server/user1 register = user2:pass...@server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer host=server username=user2 secret=passwd2 fromuser=user2 accountcode=ITCin The Asterisk CLI shows : [Oct 8 15:06:03] -- Executing [...@macro-getiaxaccount:5] MYSQL(SIP/ITCENTER-3starsnet-0764cdb0, ... [Oct 8 15:06:03] -- Executing [...@macro-getiaxaccount:6] MacroExit(SIP/ITCENTER-3starsnet-0764cdb0, ... [Oct 8 15:06:03] -- Executing [...@092:9] NoOp(SIP/ITCENTER-3starsnet-0764cdb0, ... [Oct 8 15:06:03] -- Executing [...@09:10] Dial(SIP/ITCENTER-3starsnet-0764cdb0, ... Notice the SIP/ITCENTER-3starsnet. Now when I put [ITCENTER-3starsnet] in comment in sip.conf, the CLI shows : [Oct 8 15:16:08] -- Executing [...@macro-getiaxaccount:5] MYSQL(SIP/YOCAN-3starsnet-0764e7b0, ... [Oct 8 15:16:08] -- Executing [...@macro-getiaxaccount:6] MacroExit(SIP/YOCAN-3starsnet-0764e7b0, ... [Oct 8 15:16:08] -- Executing [...@092779077:9] NoOp(SIP/YOCAN-3starsnet-0764e7b0, ... [Oct 8 15:16:08] -- Executing [...@092779077:10] Dial(SIP/YOCAN-3starsnet-0764e7b0, ... Notice the SIP/YOCAN-3starsnet. How can I keep the SIP-connection for user1 apart from the SIP-connection of user2 ??? When I activate the SIP-account for user2, an incoming call always goes via this second SIP-account !! Thanks for the feedback. Jonas. Jonas, How about breaking it up in extensions.conf. The /user1 at the end of the registration tells the device on the other end to send the call to us...@your_ip_address. You may want to try: sip.conf register = user1:pass...@server/line1 register = user2:pass...@server/line2 extensions.conf Exten = line1,1,Playback(hello) Exten = line2,1,Playback(tt-monkeys) ___ -- 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
Re: [asterisk-users] Best QoS for Linux
On Fri, Oct 9, 2009 at 2:18 AM, John A. Sullivan III jsulli...@opensourcedevel.com wrote: 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 We were thinking on similar lines a while back and decide to implement Packet Prioritization. VoIP packets to have highest priority as compared to all other packets. I believe tc, iproute2, and iptables was to be used; thou I am not very sure. Due to lack of time, we could not do this, but its still on my ToDo list. hth, Sanjay ___ -- 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] Digium G729 licence unattended install
Hi, One of the key features of Asterisk is that we can install it on many hardware platforms. We've done our best to script this installation process, so that, in case of hardware failure, we can re-install Asterisk on another platform. The question I have is how can we adapt our process so that Digium's G729 licences (or other licenced software) could be installed without asking too long interactive sessions. Before digging deeper into this topic, I guessed the installation process could be : A- install operating system on new provisioned bare-metal machine, B- install interactively Digium's G729 program, C- save relevant files on another media, D- launch unattended installation of operating system and Asterisk (using saved files). 1. Do you think it could be possible to interactively produce needed files on a provisioning server (steps B, C) or to fully script licenced software installation ? 2. We're using virtual machines to duplicate production systems for troubleshooting and development. On virtual environment such as Virtual Box, it is possible to set things like MAC address but not to set Processor speed or ID (I think virtual machines inherit many host machine characteristics). Is it then possible to buy and use, one at a time, the same licence to mimic differents production systems ? 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
[asterisk-users] Today's problem: Inbound call routing
O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium TE205P connected to a single span if ISDN PRI. The Telco has assigned us two local numbers to test incoming calls. I created an inbound route for one of those DID's and assigned it to one of our extensions. Sounds simple enough. Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower 1155 Fort Street Mall Honolulu, Hawaii 96813 Mobile: 808-782-6306 Fax: 808-533-3677 www.rolandschorr.com http://www.rolandschorr.com/ b...@rolandschorr.com ___ -- 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] MPG123 Dying
- Dovid Bender asteriskus...@dovid.net wrote: | - Original Message - | From: Trevor Peirce tpei...@digitalcon.ca | To: Asterisk Users Mailing List - Non-Commercial Discussion | asterisk-users@lists.digium.com | Sent: Tuesday, October 06, 2009 23:14 | Subject: Re: [asterisk-users] MPG123 Dying | | | --[ UxBoD ]-- wrote: | Please how do I stop the following ??? | | Asterisk ended with exit status 127 | Asterisk died with code 127. | Automatically restarting Asterisk. | mpg123: no process killed | | | You figure out why asterisk is crashing. :) | | This has nothing to do with mpg123, which is just an innocent | bystander. | | | I had an issue with mpg123 a few days ago where all of a sudden | Asterisk was | using 100% of the CPU. It happened over and over and I decided to just | | remove it. Any particular reason why you need to use mpg123 ? | On investigation it looks like a issue with my commercial Digium G729 licenses. With Asterisk CLI running I make a call, via IAX, and the following appears :- Connected to Asterisk 1.4.26.2 currently running on voip (pid = 3296) Verbosity is at least 3 -- Executing [xx:1] Dial(SIP/1001-b7d17d10, IAX2/xx/x) in new stack -- Called -- Call accepted by 217.14.138.130 (format g729) -- Format for call is g729 -- IAX2/xxx-3436 is making progress passing it to SIP/1001-b7d17d10 -- Hungup 'IAX2/x-3436' == Spawn extension (splatnix, x, 1) exited non-zero on 'SIP/1001-b7d17d10' voip*CLI Asterisk ended with exit status 127 Asterisk died with code 127. Disconnected from Asterisk server Executing last minute cleanups [r...@voip asterisk]# Automatically restarting Asterisk. Have opened a support ticket with them. -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] Today's problem: Inbound call routing
On Thu, Oct 08, 2009 at 10:00:19PM -1000, Ben Schorr wrote: O.K., so AsteriskNow 1.4.26.2, FreePBX 2.6.0RC2.1 We have a Digium TE205P connected to a single span if ISDN PRI. The Telco has assigned us two local numbers to test incoming calls. I created an inbound route for one of those DID's and assigned it to one of our extensions. Sounds simple enough. Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 That is a pretty clear error message. So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? You manage your dialplan with FreePBX. This mailing list supports Asterisk. I have no problem with questions about FreePBX systems. But they should also be phrased as Asterisk questions. This is a FreePBX question. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Asterisk Queue Agent
Hi all,
I have 2 question.
I have a call center queue with 5 agent; the following are the configuration
files:
*queue.conf*
[name_of_queue]
musicclass = default
announce = queue-name_of_queue
strategy = ringall
servicelevel = 60
context = callcenter
timeout = 60
retry = 5
wrapuptime=15
autopause=no
maxlen = 0
announce-frequency = 60
periodic-announce-frequency=30
announce-holdtime = yes
announce-round-seconds = 10
queue-youarenext=queue-youarenext ; (You are now first in line.)
queue-thereare=queue-thereare ; (There are)
queue-callswaiting=queue-callswaiting ; (calls waiting.)
queue-holdtime=queue-holdtime ; (The current est. holdtime
is)
queue-minutes=queue-minutes; (minutes.)
queue-seconds=queue-seconds; (seconds.)
queue-thankyou=queue-thankyou ; (Thank you for your
patience.)
queue-lessthan=queue-less-than ; (less than)
queue-reporthold=queue-reporthold ; (Hold time)
periodic-announce=queue-periodic-announce ; (All reps busy / wait for
next)
reportholdtime = yes
ringinuse = no
memberdelay = 3
timeoutrestart = yes
monitor-format = wav
monitor-join = no
member=agent/1,,Agent 1
member=agent/2,,Agent 2
member=agent/3,,Agent 3
member=agent/4,,Agent 4
member=agent/5,,Agent 5
*agent.conf*
[general]
persistentagents=yes
[agents]
maxlogintries=3
autologoffunavail=yes
ackcall=always
endcall=no
wrapuptime=5000
musiconhold = default
updatecdr=yes
;recordagentcalls=yes
;recordformat=wav
;urlprefix=CALLCENTER
;savecallsin=/var/calls
custom_beep=beep
agent= 1,1234,Agent 1
agent= 2,1234,Agent 2
agent= 3,1234,Agent 3
agent= 4,1234,Agent 4
agent= 5,1234,Agent 5
*FIRST QUESTION*:
if I comment in agent.conf parameters recordagentcalls I can record
conversion formed by 2 file (side in and side out) with the name that I
choose by ${MONITOR_FILENAME}, but I loose the information which agent
answer.
If I uncomment in agent.conf the parameter recordagentcalls I can view on
file name which agent answer, but I can't choose postfix file name and I
can't record the two side (in out) audio files.
Someone can help me to record two side (in out) audio name, with agent id
and a predefined postfix file name
*SECOND QUESTION*:
how can I set the queue to play an estimated hold time in queue to the
member in the queue I can play only to agent. Someone can help me
Thanks to all for your help
Marco
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Re: [asterisk-users] Digium G729 licence unattended install
On Fri, 9 Oct 2009, Olivier wrote: The question I have is how can we adapt our process so that Digium's G729 licences (or other licenced software) could be installed without asking too long interactive sessions. Download and deploy the free one. Buy digium licenses to cover each anticipated instance, but don't bother going through their long interactive registration process. Job done. (Although I do anticipate some sort of hand wringing from Digium over this ;-) Gordon ___ -- 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] G.729 and Voicemail
While we're on the subject of G.729... I can end to end use it with no transcoding, but voicemail is the main sticking point for me - I'd need to transcode. So why can't voicemail store the audio in the format it's being streamed in on? Is there a technical reason for no voicemail storage in G.729? We have prompts in G.729, so why not the messages? It doesn't have to mix anything, just store the incoming audio... Gordon ___ -- 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] Trunk and Pstn line
Hello Please let me know can we call normal PSTN lines as trunk lines?? As a normal pstn line used in home . One More thing that If i need ten PSTN lines on one Server then which Digium card is suitable. I am confused with TDM800P as it say it accepts a trunk line? -- Best Regards Shakeel Abbas ___ -- 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] How to keep difference between 2 SIP-accounts/trunks from same server ??
On Fri, Oct 9, 2009 at 10:37 AM, jonas kellens jonas.kell...@telenet.be wrote: So the call comes into the right context... that's not the problem. But my CDR is messed up. The accountcode that I have set for user1 is always replaced for the accountcode I've set for user 2. [YOCAN-3starsnet] type=peer accountcode=user1_in [ITCENTER-3starsnet] type=peer accountcode=user2_in Exactly this kind of problem I could not solved... Is there yet another workaround ?! Is it not meant to host several SIP-accounts on 1 Asterisk-box that register to a SIP- provider ??? Jonas. ... and this is my dilema also. Nini. ___ -- 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] MPG123 Dying
- Original Message - From: --[ UxBoD ]-- ux...@splatnix.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 09, 2009 10:32 Subject: Re: [asterisk-users] MPG123 Dying - Dovid Bender asteriskus...@dovid.net wrote: | - Original Message - | From: Trevor Peirce tpei...@digitalcon.ca | To: Asterisk Users Mailing List - Non-Commercial Discussion | asterisk-users@lists.digium.com | Sent: Tuesday, October 06, 2009 23:14 | Subject: Re: [asterisk-users] MPG123 Dying | | | --[ UxBoD ]-- wrote: | Please how do I stop the following ??? | | Asterisk ended with exit status 127 | Asterisk died with code 127. | Automatically restarting Asterisk. | mpg123: no process killed | | | You figure out why asterisk is crashing. :) | | This has nothing to do with mpg123, which is just an innocent | bystander. | | | I had an issue with mpg123 a few days ago where all of a sudden | Asterisk was | using 100% of the CPU. It happened over and over and I decided to just | | remove it. Any particular reason why you need to use mpg123 ? | On investigation it looks like a issue with my commercial Digium G729 licenses. With Asterisk CLI running I make a call, via IAX, and the following appears :- Connected to Asterisk 1.4.26.2 currently running on voip (pid = 3296) Verbosity is at least 3 -- Executing [xx:1] Dial(SIP/1001-b7d17d10, IAX2/xx/x) in new stack -- Called -- Call accepted by 217.14.138.130 (format g729) -- Format for call is g729 -- IAX2/xxx-3436 is making progress passing it to SIP/1001-b7d17d10 -- Hungup 'IAX2/x-3436' == Spawn extension (splatnix, x, 1) exited non-zero on 'SIP/1001-b7d17d10' voip*CLI Asterisk ended with exit status 127 Asterisk died with code 127. Disconnected from Asterisk server Executing last minute cleanups [r...@voip asterisk]# Automatically restarting Asterisk. Have opened a support ticket with them. -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration So it only happens when using G729 ? With G711U/A there is no issue ? ___ -- 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] How to keep difference between 2 SIP-accounts/trunks from same server ??
I don't think there is much you can do since Asterisk matched it based on the IP of your carrier. Maybe there is some sort of variable that you can set in the dial plan ? - Original Message - From: jonas kellens To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, October 09, 2009 09:37 Subject: Re: [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ?? What I have tried is : register = user1:pass...@server/yocan register = user2:pass...@server/itcenter extensions.conf : [default] exten = yocan,1,GoTo(user1,s,1) exten = itcenter,1,GoTo(user2,s,1) [user1] ... [user2] ... But the CLI shows : [Oct 9 09:28:52] -- Executing [...@macro-getiaxaccount:5] MYSQL(SIP/ITCENTER-3starsnet-076e4700, ... [Oct 9 09:28:52] -- Executing [...@macro-getiaxaccount:6] MacroExit(SIP/ITCENTER-3starsnet-076e4700,... [Oct 9 09:28:52] -- Executing [...@user1:9] NoOp(SIP/ITCENTER-3starsnet-076e4700, ... [Oct 9 09:28:52] -- Executing [...@user1:10] Dial(SIP/ITCENTER-3starsnet-076e4700, ... So the call comes into the right context... that's not the problem. But my CDR is messed up. The accountcode that I have set for user1 is always replaced for the accountcode I've set for user 2. [YOCAN-3starsnet] type=peer accountcode=user1_in [ITCENTER-3starsnet] type=peer accountcode=user2_in Is there yet another workaround ?! Is it not meant to host several SIP-accounts on 1 Asterisk-box that register to a SIP- provider ??? Jonas. On Thu, 2009-10-08 at 23:21 +0200, Dovid Bender wrote: - Original Message - From: jonas kellens To: Asterisk Mailing Sent: Thursday, October 08, 2009 15:20 Subject: [asterisk-users] How to keep difference between 2 SIP-accounts/trunks from same server ?? Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A -- Z. In my sip.conf : register = user1:pass...@server/user1 register = user2:pass...@server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer host=server username=user2 secret=passwd2 fromuser=user2 accountcode=ITCin The Asterisk CLI shows : [Oct 8 15:06:03] -- Executing [...@macro-getiaxaccount:5] MYSQL(SIP/ITCENTER-3starsnet-0764cdb0, ... [Oct 8 15:06:03] -- Executing [...@macro-getiaxaccount:6] MacroExit(SIP/ITCENTER-3starsnet-0764cdb0, ... [Oct 8 15:06:03] -- Executing [...@092:9] NoOp(SIP/ITCENTER-3starsnet-0764cdb0, ... [Oct 8 15:06:03] -- Executing [...@09:10] Dial(SIP/ITCENTER-3starsnet-0764cdb0, ... Notice the SIP/ITCENTER-3starsnet. Now when I put [ITCENTER-3starsnet] in comment in sip.conf, the CLI shows : [Oct 8 15:16:08] -- Executing [...@macro-getiaxaccount:5] MYSQL(SIP/YOCAN-3starsnet-0764e7b0, ... [Oct 8 15:16:08] -- Executing [...@macro-getiaxaccount:6] MacroExit(SIP/YOCAN-3starsnet-0764e7b0, ... [Oct 8 15:16:08] -- Executing [...@092779077:9] NoOp(SIP/YOCAN-3starsnet-0764e7b0, ... [Oct 8 15:16:08] -- Executing [...@092779077:10] Dial(SIP/YOCAN-3starsnet-0764e7b0, ... Notice the SIP/YOCAN-3starsnet. How can I keep the SIP-connection for user1 apart from the SIP-connection of user2 ??? When I activate the SIP-account for user2, an incoming call always goes via this second SIP-account !! Thanks for the feedback. Jonas. Jonas, How about breaking it up in extensions.conf. The /user1 at the end of the registration tells the device on the other end to send the call to us...@your_ip_address. You may want to try: sip.conf register = user1:pass...@server/line1 register = user2:pass...@server/line2 extensions.conf Exten = line1,1,Playback(hello) Exten = line2,1,Playback(tt-monkeys) ___ -- 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:
Re: [asterisk-users] MPG123 Dying
- --[ UxBoD ]-- ux...@splatnix.net wrote: | - Dovid Bender asteriskus...@dovid.net wrote: | | | - Original Message - | | From: Trevor Peirce tpei...@digitalcon.ca | | To: Asterisk Users Mailing List - Non-Commercial Discussion | | asterisk-users@lists.digium.com | | Sent: Tuesday, October 06, 2009 23:14 | | Subject: Re: [asterisk-users] MPG123 Dying | | | | | | --[ UxBoD ]-- wrote: | | Please how do I stop the following ??? | | | | Asterisk ended with exit status 127 | | Asterisk died with code 127. | | Automatically restarting Asterisk. | | mpg123: no process killed | | | | | | You figure out why asterisk is crashing. :) | | | | This has nothing to do with mpg123, which is just an innocent | | bystander. | | | | | | I had an issue with mpg123 a few days ago where all of a sudden | | Asterisk was | | using 100% of the CPU. It happened over and over and I decided to | just | | | | remove it. Any particular reason why you need to use mpg123 ? | | | On investigation it looks like a issue with my commercial Digium G729 | licenses. With Asterisk CLI running I make a call, via IAX, and the | following appears :- | | Connected to Asterisk 1.4.26.2 currently running on voip (pid = 3296) | Verbosity is at least 3 | -- Executing [xx:1] Dial(SIP/1001-b7d17d10, | IAX2/xx/x) in new stack | -- Called | -- Call accepted by 217.14.138.130 (format g729) | -- Format for call is g729 | -- IAX2/xxx-3436 is making progress passing it to | SIP/1001-b7d17d10 | -- Hungup 'IAX2/x-3436' | == Spawn extension (splatnix, x, 1) exited non-zero on | 'SIP/1001-b7d17d10' | voip*CLI Asterisk ended with exit status 127 | Asterisk died with code 127. | | Disconnected from Asterisk server | Executing last minute cleanups | [r...@voip asterisk]# Automatically restarting Asterisk. | | Have opened a support ticket with them. | Hmmm, I thought I would try and eliminate whether it was the G729 codec by trying the one from Howler. I get exactly the same error :( I place a IAX call, which is fine, but as soon as I hangup it kills Asterisk. Any pointers on how to debug this please ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] MPG123 Dying
- Dovid Bender asteriskus...@dovid.net wrote: | - Original Message - | From: --[ UxBoD ]-- ux...@splatnix.net | To: Asterisk Users Mailing List - Non-Commercial Discussion | asterisk-users@lists.digium.com | Sent: Friday, October 09, 2009 10:32 | Subject: Re: [asterisk-users] MPG123 Dying | | | - Dovid Bender asteriskus...@dovid.net wrote: | | | - Original Message - | | From: Trevor Peirce tpei...@digitalcon.ca | | To: Asterisk Users Mailing List - Non-Commercial Discussion | | asterisk-users@lists.digium.com | | Sent: Tuesday, October 06, 2009 23:14 | | Subject: Re: [asterisk-users] MPG123 Dying | | | | | | --[ UxBoD ]-- wrote: | | Please how do I stop the following ??? | | | | Asterisk ended with exit status 127 | | Asterisk died with code 127. | | Automatically restarting Asterisk. | | mpg123: no process killed | | | | | | You figure out why asterisk is crashing. :) | | | | This has nothing to do with mpg123, which is just an innocent | | bystander. | | | | | | I had an issue with mpg123 a few days ago where all of a sudden | | Asterisk was | | using 100% of the CPU. It happened over and over and I decided to | just | | | | remove it. Any particular reason why you need to use mpg123 ? | | | On investigation it looks like a issue with my commercial Digium | G729 | licenses. With Asterisk CLI running I make a call, via IAX, and the | | following appears :- | | Connected to Asterisk 1.4.26.2 currently running on voip (pid = | 3296) | Verbosity is at least 3 | -- Executing [xx:1] Dial(SIP/1001-b7d17d10, | IAX2/xx/x) in new stack | -- Called | -- Call accepted by 217.14.138.130 (format g729) | -- Format for call is g729 | -- IAX2/xxx-3436 is making progress passing it to | SIP/1001-b7d17d10 | -- Hungup 'IAX2/x-3436' | == Spawn extension (splatnix, x, 1) exited non-zero on | | 'SIP/1001-b7d17d10' | voip*CLI Asterisk ended with exit status 127 | Asterisk died with code 127. | | Disconnected from Asterisk server | Executing last minute cleanups | [r...@voip asterisk]# Automatically restarting Asterisk. | | Have opened a support ticket with them. | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | So it only happens when using G729 ? With G711U/A there is no issue ? Good point ... switched to ulaw and the same issue arises ... Ran IAX debug but that did not really show anything :( Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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
We use 3Com managed gigabit switches that support QoS and priority for VoIP. 3Com Unified Gigabit Wireless PoE Switch 24 and 3Com Baseline Switch 2924-PWR Plus Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Michelle Dupuis wrote: 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? ___ -- 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
Re: [asterisk-users] MeetMe option question
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Kenner
Sent: Thursday, October 08, 2009 8:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MeetMe option question
We've started to use Asterisk for conferencing and have been getting
some
complaints. Our configuration is that some people call in from home,
but
we have a physical conference room with a Polycom. When somebody was
giving
a presentation in the physical conference room, we were told that the
remote
people kept hearing him cut in and our. To me, this sounds like the
talking
optimization was getting false negatives.
Is there a way to say don't apply talk optimization to this user so we
could add that to the Polycom when it called it? From a quick scan of
app_meetme.c, I don't see one, but it doesn't look too hard to add.
--
You can do this in the dialplan. Just launch MeetMe with different
options based on the caller, I use SQL and AGI in my installations but
it doesn't have to be that complex.
If (${CALLERID(num)} = Polycom callerID) {
MeetMe(CONFROOM|AscM);
} else {
MeetMe(CONFROOM|AscMo);
}
My syntax is probably off a bit but that should get you started. You may
also want to consider just turning off the talker optimization entirely
- I've found it to be very problematic and generates more complaints
than it's worth.
Bob
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Re: [asterisk-users] No sound on voicemail from analog line
--- On Thu, 10/8/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: From: Tzafrir Cohen tzafrir.co...@xorcom.com Subject: Re: [asterisk-users] No sound on voicemail from analog line To: asterisk-users@lists.digium.com Date: Thursday, October 8, 2009, 4:11 PM On Thu, Oct 08, 2009 at 12:43:00PM -0700, Landy Landy wrote: Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What do you mean by voicemail from PSTN? Asterisk's voicemail or the provider's ? The cards is FXS? FXO? T1? E1? Well, what I mean is on calls coming in from outside on the analog line. The card is one of those old modems X100p, I guess is a clone card. ___ -- 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] wrond DTMF detection on Zap channel
Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue 2) wait 5 seconds if the queue is empty 3) pick-up a call from the queue and transfer it to an extension othervise Finally, the extension lanuch another AGI php script that requires some DTMF tone to the user to perform some actions. This system is working properly since 2006. Well, the problem during last days is that it seems that sometimes the DTMF recognition doesn't work, in the debug i get: AGI Tx 200 result=0 But users complains to me because they assure to have digited something different than 0. The problem seems to be reproducible when the system is loaded (i don't have information on the SO but we receive abut 2500 calls per hour each call is very short because usually it is hangup after a very short time, as the queue length is very often 1) It's not an AGI application problem as i get the wrong dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*/ nik600 http://www.kumbe.it ___ -- 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] g729 free codec any idea
Before we call each other liars and thieves, here is a link: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ As with any open source, do your own homework on licensing, AND apply your own reasonable judgment. At this point I would like to confess that I watch rented DVD's under Linux...which is illegal since there is no license for decrypting commercial DVD's under Linux. I'm overwhelmed with guilt and feel better having that off my chest. As the sparks fly I'll happily walk away from this thread :) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Thursday, October 08, 2009 10:47 PM To: Asterisk Users List Subject: Re: [asterisk-users] g729 free codec any idea On 9/10/09 3:31 PM, Michelle Dupuis wrote: I believe that Intel placed a 729 codec into the public domain (free), and someone wrapped it in a nice Asterisk package for use. No idea where - but I do recall that it is out there, and legal. Of course it's nice to support a vendor, but free alternatives can't be shunned... The original comment stands. The codec is patented. The implementation is not. In order to use the implementation you need a license unless you live somewhere that: A) Doesn't have patents B) Doesn't have a trade agreement with USA Inserting a g729 codec from a licensed source other than Digium will break the GPL (Digium issues an exception for it's g729): http://tinyurl.com/ykpu42u This conversation has come up hundreds of times on this mailing list and the result is always the same - if you're happy breaking the law, go for it - if you get most of your movies from piratebay then it probably isn't a problem for you. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
Re: [asterisk-users] MPG123 Dying
- Dovid Bender asteriskus...@dovid.net wrote: | - Original Message - | From: --[ UxBoD ]-- ux...@splatnix.net | To: Asterisk Users Mailing List - Non-Commercial Discussion | asterisk-users@lists.digium.com | Sent: Friday, October 09, 2009 10:32 | Subject: Re: [asterisk-users] MPG123 Dying | | | - Dovid Bender asteriskus...@dovid.net wrote: | | | - Original Message - | | From: Trevor Peirce tpei...@digitalcon.ca | | To: Asterisk Users Mailing List - Non-Commercial Discussion | | asterisk-users@lists.digium.com | | Sent: Tuesday, October 06, 2009 23:14 | | Subject: Re: [asterisk-users] MPG123 Dying | | | | | | --[ UxBoD ]-- wrote: | | Please how do I stop the following ??? | | | | Asterisk ended with exit status 127 | | Asterisk died with code 127. | | Automatically restarting Asterisk. | | mpg123: no process killed | | | | | | You figure out why asterisk is crashing. :) | | | | This has nothing to do with mpg123, which is just an innocent | | bystander. | | | | | | I had an issue with mpg123 a few days ago where all of a sudden | | Asterisk was | | using 100% of the CPU. It happened over and over and I decided to | just | | | | remove it. Any particular reason why you need to use mpg123 ? | | | On investigation it looks like a issue with my commercial Digium | G729 | licenses. With Asterisk CLI running I make a call, via IAX, and the | | following appears :- | | Connected to Asterisk 1.4.26.2 currently running on voip (pid = | 3296) | Verbosity is at least 3 | -- Executing [xx:1] Dial(SIP/1001-b7d17d10, | IAX2/xx/x) in new stack | -- Called | -- Call accepted by 217.14.138.130 (format g729) | -- Format for call is g729 | -- IAX2/xxx-3436 is making progress passing it to | SIP/1001-b7d17d10 | -- Hungup 'IAX2/x-3436' | == Spawn extension (splatnix, x, 1) exited non-zero on | | 'SIP/1001-b7d17d10' | voip*CLI Asterisk ended with exit status 127 | Asterisk died with code 127. | | Disconnected from Asterisk server | Executing last minute cleanups | [r...@voip asterisk]# Automatically restarting Asterisk. | | Have opened a support ticket with them. | | -- | This message has been scanned for viruses and | dangerous content and is believed to be clean. | | SplatNIX IT Services :: Innovation through collaboration | | | So it only happens when using G729 ? With G711U/A there is no issue ? | | Incase anybody else hits this ! http://bugs.centos.org/view.php?id=3889 -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] G.729 and Voicemail
Gordon Henderson wrote: While we're on the subject of G.729... I can end to end use it with no transcoding, but voicemail is the main sticking point for me - I'd need to transcode. So why can't voicemail store the audio in the format it's being streamed in on? Why do you think it can't? It most certainly can, but only if that's the only format you will get connections in... if all your calls are G.729, just configure voicemail.conf to store in g729 format. If your calls are in a mixture of formats, then you'll have to use transcoding, but that's unavoidable (if the voicemail was left via a G.729 call and stored that way, but then later someone wants to listen to it using a G.711 call, then it will still have to be transcoded). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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 option question
Robert McGilvray wrote: You can do this in the dialplan. Just launch MeetMe with different options based on the caller, What's confusing me is that when I look in app_meetme.c, the relevant options are stored in what are called conference flags and there are separate user flags. This makes it confusing as to which (if any?) are global for the conference and which can be set per-user. ___ -- 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 option question
Richard Kenner ken...@gnat.com wrote: Robert McGilvray wrote: You can do this in the dialplan. Just launch MeetMe with different options based on the caller, What's confusing me is that when I look in app_meetme.c, the relevant options are stored in what are called conference flags and there are separate user flags. This makes it confusing as to which (if any?) are global for the conference and which can be set per-user. Have you tried without that option -- maybe it would not make much difference. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- 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] VoiceMail and IMAP
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 ??? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration ___ -- 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] g729 free codec any idea
On Fri, 9 Oct 2009, Michelle Dupuis wrote: Before we call each other liars and thieves, here is a link: http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ As with any open source, do your own homework on licensing, AND apply your own reasonable judgment. At this point I would like to confess that I watch rented DVD's under Linux...which is illegal since there is no license for decrypting commercial DVD's under Linux. I'm overwhelmed with guilt and feel better having that off my chest. As the sparks fly I'll happily walk away from this thread :) If you read the entire page of what you are linking to above you will see that it verifies what everyone has been saying all along - it is NOT FREE. If you want to use it you are required to pay royalties. You said it was free. It is not. j ___ -- 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] G.729 and Voicemail
On Fri, 9 Oct 2009, Kevin P. Fleming wrote: Gordon Henderson wrote: While we're on the subject of G.729... I can end to end use it with no transcoding, but voicemail is the main sticking point for me - I'd need to transcode. So why can't voicemail store the audio in the format it's being streamed in on? Why do you think it can't? It most certainly can, but only if that's the only format you will get connections in... OK - You've got me there. I did try this once upon a time and it didn't work, and the config file (and wiki) seemed to imply that only the 3 formats there were supported. Maybe I didn't try hard enough! if all your calls are G.729, just configure voicemail.conf to store in g729 format. If your calls are in a mixture of formats, then you'll have to use transcoding, but that's unavoidable (if the voicemail was left via a G.729 call and stored that way, but then later someone wants to listen to it using a G.711 call, then it will still have to be transcoded). All deskphnoes I've ever bought support g729 natively. They also all support G711. The wholesale termination services I use all support g729 too. The only fly in the oinkment is local PSTN connections which are obviously g711a. Now if voicemail would just blindly store in the incoming format, then that would be nice - VM files would be stored in g729 or g711, or whatever else comes in. It's just data afterall - why even think about transcoding? (Although I'm sure the answer will be something to do with trying to play g729, etc. attachments from an email or switching codecs mid-call from playing the prompts to playing the message) It's all a bit bothersome - especially when dealing with weedy processors which barely manage a few GSM transcodes )-: Gordon ___ -- 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] 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
[asterisk-users] Billing applications
Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. True CDR, ___ -- 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] Billing applications
Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- 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] Billing applications
A2billing (Star2Billing, I think, for commercial support) is a good choice and it's very mature software. Astercc is very fast and has a very good callshop solution. Regards, Juan voip crazy wrote: Hello all, I want to instal a Billing solution in the same asterisk's box. I have browse for ast2bill asterisk billing, astercc, and more, bu ti do not know which will be the best for me. The only things i need, are, - Postpaid and prepaid applications. - True CDR. Better that asterisk one, With suport for transfers - I do not need support for reseller - Billing for Voip, PSTN trunks I need a light app. I'm not searching a heavy app. with a lots of modules and applicacions. I need a ligth application for a soho and its needs. Any one are using a billing application which fits this needs? Any clue will be welcomed. Thanks in advance. VoipCrazy ___ -- 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
Re: [asterisk-users] G.729 and Voicemail
Gordon Henderson wrote: All deskphnoes I've ever bought support g729 natively. They also all support G711. The wholesale termination services I use all support g729 too. The only fly in the oinkment is local PSTN connections which are obviously g711a. Now if voicemail would just blindly store in the incoming format, then that would be nice - VM files would be stored in g729 or g711, or whatever else comes in. It's just data afterall - why even think about transcoding? (Although I'm sure the answer will be something to do with trying to play g729, etc. attachments from an email or switching codecs mid-call from playing the prompts to playing the message) You missed my point; let's say a call comes in over your PSTN connection, goes to voicemail, and they leave a voicemail, in G.711a format. Then, one of your internal users places a call to the voicemail system from their G.729 phone to retrieve their messages; now for that message to be played back, it must be sent to the phone in G.729 format, because that's what format the phone and Asterisk chose when the call was placed. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] calls ansowered for 1 second or less
Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, its just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : vendor SIP--à Asterisk ßIAX2---à Asterisk with Dhadi channels Here: -- Executing [966505103...@from-internal:1] Macro(SIP/100-b609f9c0, user-callerid|SKIPTTL|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/100-b609f9c0, AMPUSER=100) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/100-b609f9c0, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/100-b609f9c0, 1|Set|REALCALLERIDNUM=100) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/100-b609f9c0, AMPUSER=100) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/100-b609f9c0, AMPUSERCIDNAME=100) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/100-b609f9c0, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/100-b609f9c0, AMPUSERCID=100) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/100-b609f9c0, CALLERID(all)=100 100) in new stack -- Executing [...@macro-user-callerid:9] Set(SIP/100-b609f9c0, REALCALLERIDNUM=100) in new stack -- Executing [...@macro-user-callerid:10] ExecIf(SIP/100-b609f9c0, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:11] GotoIf(SIP/100-b609f9c0, 1?continue) in new stack -- Goto (macro-user-callerid,s,20) -- Executing [...@macro-user-callerid:20] NoOp(SIP/100-b609f9c0, Using CallerID 100 100) in new stack -- Executing [966505103...@from-internal:2] Set(SIP/100-b609f9c0, _NODEST=) in new stack -- Executing [966505103...@from-internal:3] Macro(SIP/100-b609f9c0, record-enable|100|OUT|) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/100-b609f9c0, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/100-b609f9c0, recordingcheck|20091009-194302|1255102982.3126) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20091009-194302|1255102982.3126: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/100-b609f9c0, ) in new stack -- Executing [966505103...@from-internal:4] Macro(SIP/100-b609f9c0, dialout-trunk|12|505103150||) in new stack -- Executing [...@macro-dialout-trunk:1] Set(SIP/100-b609f9c0, DIAL_TRUNK=12) in new stack -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/100-b609f9c0, 0?sub-pincheck|s|1) in new stack -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/100-b609f9c0, 0?disabletrunk|1) in new stack -- Executing [...@macro-dialout-trunk:4] Set(SIP/100-b609f9c0, DIAL_NUMBER=505103150) in new stack -- Executing [...@macro-dialout-trunk:5] Set(SIP/100-b609f9c0, DIAL_TRUNK_OPTIONS=trf) in new stack -- Executing [...@macro-dialout-trunk:6] Set(SIP/100-b609f9c0, OUTBOUND_GROUP=OUT_12) in new stack -- Executing [...@macro-dialout-trunk:7] GotoIf(SIP/100-b609f9c0, 1?nomax) in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [...@macro-dialout-trunk:9] GotoIf(SIP/100-b609f9c0, 0?skipoutcid) in new stack -- Executing [...@macro-dialout-trunk:10] Set(SIP/100-b609f9c0, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [...@macro-dialout-trunk:11] Macro(SIP/100-b609f9c0, outbound-callerid|12) in new stack -- Executing [...@macro-outbound-callerid:1] ExecIf(SIP/100-b609f9c0, 0|SetCallerPres|) in new stack -- Executing [...@macro-outbound-callerid:2] ExecIf(SIP/100-b609f9c0, 0|Set|REALCALLERIDNUM=100) in new stack -- Executing [...@macro-outbound-callerid:3] GotoIf(SIP/100-b609f9c0, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [...@macro-outbound-callerid:6] Set(SIP/100-b609f9c0, USEROUTCID=) in new stack -- Executing [...@macro-outbound-callerid:7] Set(SIP/100-b609f9c0, EMERGENCYCID=) in new stack -- Executing [...@macro-outbound-callerid:8] Set(SIP/100-b609f9c0, TRUNKOUTCID=9) in new stack -- Executing [...@macro-outbound-callerid:9] GotoIf(SIP/100-b609f9c0, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [...@macro-outbound-callerid:12] ExecIf(SIP/100-b609f9c0, 1|Set|CALLERID(all)=9) in new stack -- Executing [...@macro-outbound-callerid:13] GotoIf(SIP/100-b609f9c0, 1?exit) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing [...@macro-outbound-callerid:11] MacroExit(SIP/100-b609f9c0, ) in new stack -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/100-b609f9c0, 0|AGI|fixlocalprefix) in new stack -- Executing [...@macro-dialout-trunk:13] Set(SIP/100-b609f9c0, OUTNUM=0505103150) in new stack
Re: [asterisk-users] Today's problem: Inbound call routing
-Original Message- Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 That is a pretty clear error message. Yes, I thought so. But how do I fix it? So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? You manage your dialplan with FreePBX. This mailing list supports Asterisk. I have no problem with questions about FreePBX systems. But they should also be phrased as Asterisk questions. This is a FreePBX question. I see, so this isn't an Asterisk problem it's a FreePBX problem? Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com ___ -- 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] G.729 and Voicemail
On Fri, 9 Oct 2009, Kevin P. Fleming wrote: Gordon Henderson wrote: All deskphnoes I've ever bought support g729 natively. They also all support G711. The wholesale termination services I use all support g729 too. The only fly in the oinkment is local PSTN connections which are obviously g711a. Now if voicemail would just blindly store in the incoming format, then that would be nice - VM files would be stored in g729 or g711, or whatever else comes in. It's just data afterall - why even think about transcoding? (Although I'm sure the answer will be something to do with trying to play g729, etc. attachments from an email or switching codecs mid-call from playing the prompts to playing the message) You missed my point; let's say a call comes in over your PSTN connection, goes to voicemail, and they leave a voicemail, in G.711a format. Then, one of your internal users places a call to the voicemail system from their G.729 phone to retrieve their messages; now for that message to be played back, it must be sent to the phone in G.729 format, because that's what format the phone and Asterisk chose when the call was placed. No - I understood what you were saying, but maybe I should have said that all deskphones undesrstand many codecs, not just g729. (E.g. Snom, Grandstream, etc. all support many codecs and I'm not going to criple them by turning them off) So when on a PSTN call, they speak g711, when on a VoIP trunk call they speak g729 - when an incoming VoIP call goes to voicemail it records in whatever format the incoming trunk is in, so when the deskphone plays it back, it gets the file in whatever format it's stored in and can play it. However, I don't think that is going to work for the reasons I said above, (changing codecs during one call) although it would be nice if it did. It's basically a PITA when mixing PSTN and VoIP. Fortunately a lot of my customers are VoIP only, so now knowing that I can store voicemail in g729 format, I'll go off and try that and save myself some bandwidth. Gordin ___ -- 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] G.729 and Voicemail
On Fri, 9 Oct 2009, Gordon Henderson wrote: On Fri, 9 Oct 2009, Kevin P. Fleming wrote: Gordon Henderson wrote: All deskphnoes I've ever bought support g729 natively. They also all support G711. The wholesale termination services I use all support g729 too. The only fly in the oinkment is local PSTN connections which are obviously g711a. Now if voicemail would just blindly store in the incoming format, then that would be nice - VM files would be stored in g729 or g711, or whatever else comes in. It's just data afterall - why even think about transcoding? (Although I'm sure the answer will be something to do with trying to play g729, etc. attachments from an email or switching codecs mid-call from playing the prompts to playing the message) You missed my point; let's say a call comes in over your PSTN connection, goes to voicemail, and they leave a voicemail, in G.711a format. Then, one of your internal users places a call to the voicemail system from their G.729 phone to retrieve their messages; now for that message to be played back, it must be sent to the phone in G.729 format, because that's what format the phone and Asterisk chose when the call was placed. No - I understood what you were saying, but maybe I should have said that all deskphones undesrstand many codecs, not just g729. (E.g. Snom, Grandstream, etc. all support many codecs and I'm not going to criple them by turning them off) So when on a PSTN call, they speak g711, when on a VoIP trunk call they speak g729 - I would be very surprised if that were true. Your phones speak many codecs, but they negotiate with asterisk on registration which one they will be using. They don't switch codecs based on the remote channel (which they don't even know about). Today, if your phones are negotiating 729 on registration, you are definitely transcoding calls to/from the PSTN. when an incoming VoIP call goes to voicemail it records in whatever format the incoming trunk is in, so when the deskphone plays it back, it gets the file in whatever format it's stored in and can play it. Again the phone won't switch negotiated codecs to match the file - transcoding will simply take place if they don't match. However, I don't think that is going to work for the reasons I said above, (changing codecs during one call) although it would be nice if it did. It's basically a PITA when mixing PSTN and VoIP. Fortunately a lot of my customers are VoIP only, so now knowing that I can store voicemail in g729 format, I'll go off and try that and save myself some bandwidth. Thats ounds like the best plan! j Gordin ___ -- 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
Re: [asterisk-users] wrond DTMF detection on Zap channel
are you using chan_local? try disabling the hardware DTMF. Sent using my wired Blueberry. On 10/9/09, nik600 nik...@gmail.com wrote: Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue 2) wait 5 seconds if the queue is empty 3) pick-up a call from the queue and transfer it to an extension othervise Finally, the extension lanuch another AGI php script that requires some DTMF tone to the user to perform some actions. This system is working properly since 2006. Well, the problem during last days is that it seems that sometimes the DTMF recognition doesn't work, in the debug i get: AGI Tx 200 result=0 But users complains to me because they assure to have digited something different than 0. The problem seems to be reproducible when the system is loaded (i don't have information on the SO but we receive abut 2500 calls per hour each call is very short because usually it is hangup after a very short time, as the queue length is very often 1) It's not an AGI application problem as i get the wrong dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*/ nik600 http://www.kumbe.it ___ -- 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
Re: [asterisk-users] G.729 and Voicemail
I would be very surprised if that were true. Your phones speak many codecs, but they negotiate with asterisk on registration which one they will be using. They don't switch codecs based on the remote channel (which they don't even know about). Today, if your phones are negotiating 729 on registration, you are definitely transcoding calls to/from the PSTN. Assuming we're talking about SIP (and any other voip protocol I know of for that matter), that is incorrect, codec negotiation is done during the call setup based on preferences stored in both Asterisk and the phone (that is what the SDP is for among other things). However at that point Asterisk does not know that the dial plan is going to call the voicemail application to play a file in g729 format (how can possibly now that), and therefore when the file is being played the phone already expects the audio for the call in the format negotiated during call setup which may or may not be g729. Not sure if a re-invite could be issued to change the codec type in the middle of the call, but I suppose it should be possible to implement. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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] G.729 and Voicemail
On Fri, 9 Oct 2009, Moises Silva wrote: I would be very surprised if that were true. Your phones speak many codecs, but they negotiate with asterisk on registration which one they will be using. They don't switch codecs based on the remote channel (which they don't even know about). Today, if your phones are negotiating 729 on registration, you are definitely transcoding calls to/from the PSTN. Assuming we're talking about SIP (and any other voip protocol I know of for that matter), that is incorrect, codec negotiation is done during the call setup based on preferences stored in both Asterisk and the phone (that is what the SDP is for among other things). However at that point Asterisk does not know that the dial plan is going to call the voicemail application to play a file in g729 format (how can possibly now that), and therefore when the file is being played the phone already expects the audio for the call in the format negotiated during call setup which may or may not be g729. Not sure if a re-invite could be issued to change the codec type in the middle of the call, but I suppose it should be possible to implement. Of course you are correct - not during registration but during call setup. The main idea, though, is that the negotiation doesn't change based on the far leg of the call inbound or outbound. If the phone is set to negiotiate G729 it will always negotiate G729 regardless of the far end. j -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- 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] How to keep difference between 2 SIP-accounts/trunks from same server ??
Well, in the dialplan I now use Set(CDR(accountcode)=...) in the context of an incoming call. It looks like this works for me. So I can keep track of which account is receiving which call... and thus separating them. On Fri, 2009-10-09 at 13:19 +0200, Dovid Bender wrote: I don't think there is much you can do since Asterisk matched it based on the IP of your carrier. Maybe there is some sort of variable that you can set in the dial plan ? ___ -- 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] Chanspy
How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(801) exten = 501**,n,Hangup exten = 502**,1,Chanspy(802) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. Thanks. Torintino _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009___ -- 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] Chanspy
Use ExtenSpy for spying on a specific extension. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ExtenSpy On 9 Oct, 2009, at 10:44 AM, Torintino T wrote: How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(801) exten = 501**,n,Hangup exten = 502**,1,Chanspy(802) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. Thanks. Torintino Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. ATT1.c ___ -- 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] Chanspy
Edited the post How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 502**,1,Chanspy(502) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. and is there any option for spying on a dedicated queue? Thanks. Torintino Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010___ -- 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] No sound on voicemail from analog line
Landy Landy wrote: Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound Do you mean that incoming calls on your PSTN line works as they should, but not when they reach the voicemail? or that incomming calls on PSTN are always mute? -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- 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] Today's problem: Inbound call routing
On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote: -Original Message- Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 That is a pretty clear error message. Yes, I thought so. But how do I fix it? So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? You manage your dialplan with FreePBX. This mailing list supports Asterisk. I have no problem with questions about FreePBX systems. But they should also be phrased as Asterisk questions. This is a FreePBX question. I see, so this isn't an Asterisk problem it's a FreePBX problem? Creating an inbound route is FreePBX speak. This is a FreePBX question. Please ask an Asterisk question. For instance, show a dialplan trace, show the respective dialplan, show the respective channel configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Chanspy
I think this is a dialplan problem. I would code it this way: [chanspy] include = chanspy-custom exten = 5010,1,Chanspy(501) exten = 5010,n,Hangup exten = 5020,1,Chanspy(502) Exten = 5020,n,Hangup That way 5010 would spy on 501 and 5020 would spy on 502. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Friday, October 09, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Chanspy Edited the post How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 502**,1,Chanspy(502) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. and is there any option for spying on a dedicated queue? Thanks. Torintino _ Windows Live Hotmail: Your http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 friends can get your Facebook updates, right from HotmailR. _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/so cial-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010 you. ___ -- 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] ${REASON} not getting set.
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm told
should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the callfile:
Channel: local/1
Callerid:Tests 1
MaxRetries: 0
RetryTime: 1
WaitTime: 90
Account: Dialout
Context: Diaout
Extension: dialer
Priority: 1
According to the wiki, this should work.
Here is my failed extension:
exten = failed,1,noop( Call to ${phone_number} failed ${reason}
The call to noop works and this is what I see on the console:
-- Executing [fai...@dialout:1] NoOp(OutgoingSpoolFailed, Call to
1 failed) in new stack
== Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN'
[Oct 9 14:16:50] NOTICE[29172]: pbx_spool.c:341 attempt_thread: Call failed
to go through, reason 0
[Oct 9 14:16:50] WARNING[15708]: pbx_spool.c:245 apply_outgoing: At least one
of app or extension must be specified, along with tech and dest in
file /var/spool/asterisk/outgoing/t.call
Does this just not work? Or am I missing something?
Thanks in advance,
--
Take care and have fun,
Mike Diehl.
___
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Re: [asterisk-users] Today's problem: Inbound call routing
Sorry, I'm brand new at Asterisk (and/or FreePBX). I'm going to have to figure out what all those things are before I can show them. I'll have to get back to you. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, October 09, 2009 9:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Today's problem: Inbound call routing On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote: -Original Message- Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 That is a pretty clear error message. Yes, I thought so. But how do I fix it? So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? You manage your dialplan with FreePBX. This mailing list supports Asterisk. I have no problem with questions about FreePBX systems. But they should also be phrased as Asterisk questions. This is a FreePBX question. I see, so this isn't an Asterisk problem it's a FreePBX problem? Creating an inbound route is FreePBX speak. This is a FreePBX question. Please ask an Asterisk question. For instance, show a dialplan trace, show the respective dialplan, show the respective channel configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
Re: [asterisk-users] ${REASON} not getting set.
I believe it may be because you have not told what context the local channel
should use. Try using:
Channel: local/15...@mycontext
Obviously change the mycontext to the name of the context that you want to use.
That may work for you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Saturday, 10 October 2009 6:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ${REASON} not getting set.
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm told
should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the callfile:
Channel: local/1
Callerid:Tests 1
MaxRetries: 0
RetryTime: 1
WaitTime: 90
Account: Dialout
Context: Diaout
Extension: dialer
Priority: 1
According to the wiki, this should work.
Here is my failed extension:
exten = failed,1,noop( Call to ${phone_number} failed ${reason}
The call to noop works and this is what I see on the console:
-- Executing [fai...@dialout:1] NoOp(OutgoingSpoolFailed, Call to
1 failed) in new stack
== Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN'
[Oct 9 14:16:50] NOTICE[29172]: pbx_spool.c:341 attempt_thread: Call failed
to go through, reason 0
[Oct 9 14:16:50] WARNING[15708]: pbx_spool.c:245 apply_outgoing: At least one
of app or extension must be specified, along with tech and dest in
file /var/spool/asterisk/outgoing/t.call
Does this just not work? Or am I missing something?
Thanks in advance,
--
Take care and have fun,
Mike Diehl.
___
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Register Now: http://www.astricon.net
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[asterisk-users] choppy sound
Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? 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
Re: [asterisk-users] choppy sound
It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? 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
Re: [asterisk-users] ${REASON} not getting set.
I've tried fourvariations on this theme:
Channel: local/15...@default
Channel: local/15...@dialout
Channel: local/1/default
Channel: local/1/Dialout
Neither one worked. I appreciate your time. Any other ideas?
Mike.
P.S. I thought that setting the context in the callfile took care of the
issue you suggested?
On Friday 09 October 2009 02:44:00 pm David Klaverstyn wrote:
I believe it may be because you have not told what context the local
channel should use. Try using:
Channel: local/15...@mycontext
Obviously change the mycontext to the name of the context that you want to
use.
That may work for you.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Saturday, 10 October 2009 6:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] ${REASON} not getting set.
Hi all,
I've got a program that creates a callfile and most if it working great.
However, when a call fails, I'm trying to capture the reason, which I'm
told should be in the ${REASON} channel variable.
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Here is an excerpt from the callfile:
Channel: local/1
Callerid:Tests 1
MaxRetries: 0
RetryTime: 1
WaitTime: 90
Account: Dialout
Context: Diaout
Extension: dialer
Priority: 1
According to the wiki, this should work.
Here is my failed extension:
exten = failed,1,noop( Call to ${phone_number} failed ${reason}
The call to noop works and this is what I see on the console:
-- Executing [fai...@dialout:1] NoOp(OutgoingSpoolFailed, Call
to 1 failed) in new stack
== Auto fallthrough, channel 'OutgoingSpoolFailed' status is 'UNKNOWN'
[Oct 9 14:16:50] NOTICE[29172]: pbx_spool.c:341 attempt_thread: Call
failed to go through, reason 0
[Oct 9 14:16:50] WARNING[15708]: pbx_spool.c:245 apply_outgoing: At least
one of app or extension must be specified, along with tech and dest in file
/var/spool/asterisk/outgoing/t.call
Does this just not work? Or am I missing something?
Thanks in advance,
--
Take care and have fun,
Mike Diehl.
___
-- 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
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[asterisk-users] Incoming extension not working.
Hi, all. I'm probably doing Something Dumb(tm), so please feel free to point out whatever I'm missing, no matter how stupid. Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get: Rejected connect attempt from 64.2.142.19, who was trying to reach '6031234567@' This leads me to my first question -- why doesn't it show a context? (My second is, what's wrong with the snippets, below?): iax.conf: [vitelity] context=vitelity register = username:passw...@inbound6.vitelity.net extensions.conf: [vitelity] ; Figured I'd try both things usually used to answer... exten = 6034713217,1,Answer exten = s,1,Answer [...] [default] include = vitelity Thanks... -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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
Hi, I am using CentOS Asterisk 1.4 The server has 4GB RAM, 2Ghz Duo Core, and digium 24ports fxo no hardware echo cancelation Does hardware echo will help? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? 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
Re: [asterisk-users] choppy sound
By the way, how to schedule auto reboot? thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? 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
[asterisk-users] lawnmower man attack sip tag=Zerogij34 some one else notice this in 20th september or recently?
Dear all, According to: http://www.honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/ Citibankhas been under a telephone calling attack in 20th september. Does anyone in asterisk community got any CDRs or logging of similar attacks as the one above mentioned ? Any one with logging of it or future information about the case ? Identified more detaills in this attack ? Citibank is or has been under a telephone calling attack latest 12 hours. Here I will explain the attack and how it was done. Have you seen the movie “lawnmower man”, when in the end, all phones rings in the who city? This was the aim for todays attack on Citibank in UK. The attack was simple, but probably effective when it was active. Send SIP INVITE to open SIP gateways and PBXs, who then will actually use the traditional phonesystem (POTS) to call the target. Suddenly you need DoS protection on your traditional POTS lines…. The SIP INVITE looks like this. INVITE sip:00442075005...@x SIP/2.0 Via: SIP/2.0/UDP 217.23.7.47:58585;branch=z9hG4bKaergjerugroijrgrg To: sip:x From: sip:217.23.7.47:58585;tag=Zerogij34 Call-ID: 213948958-34384780214-384...@217.23.7.47 CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:s...@217.23.7.47:58585;transport=udp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 520 Session-Expires: 3600; Allow-Events: refer.. v=0 o=sip 2147483647 1 IN IP4 1.1.1.1 s=sip c=IN IP4 1.1.1.1 t=0 0 m=audio 29784 RTP/AVP 8 0 4 18 18 18 18 96 3 98 a=rtpmap:96 telephone-event/8000 a=sendrecva=ptime:20 a=rtpmap:18 G729AB/8000 a=rtpmap:18 G729B/8000 a=rtpmap:18 G729A/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723 Lets walk through the SIP packet and see what info we can get from it: A quick google search on the tag: Zerogij34 reveals that this attack has been around since at least 6th of August. The IP (217.23.7.47)from this packet should be located in Portugal but the other attacks originate from both UK and Netherlands. There is no User-Agent listed, so the packet is very likely crafted from toosl like sipsak or sipp. The codec list seems real, but they use an obscure address (1.1.1.1) for the RTP. If they would use their own IP address, it could case a small DoS with RTP traffic for every successful call.)The port 29784 is within the range of Cisco units (26 000-32 000) The other INVITES reveals that the attacker is trying to figure the extension to get a dial-tone: * INVITE sip:00442075005...@67.170.104.216 SIP/2.0 * INVITE sip:011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:900442075005...@67.170.104.216 SIP/2.0 * INVITE sip:9011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:90442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * and several more… But is this a DoS attack on Citibank? I doubt it. Why call the Citibank on a Sunday 5 a.m.? This is more likely that Citibank has lots of lines and therefore the SIP INVITES does not generate an error (busy or others). The attacker does not hear any ringtone, but he/she should see the 180 Ringing / 180 Session in Progress. Then he or she knows that he could actually get through to the PSTN on this SIP proxy. If it would be a ringing attack, why does the attacker just send one single SIP INVITE through each gateway that actually calls this destination? The machines with the attacking IP addresses should be put under surveillance to see who connects to these. They are probably just some bots in a larger network, but they need to relay back which gateways actually responded successfully. Sad to say, but I believe this is only the small beginning…. Looking forward to hearing from you guys ;) Cheers, -- Marco Mouta ___ -- 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] lawnmower man attack ??
Dear all, According to: w w w .honeynor.no/2009/09/20/citibank-uk-number-was-target-for-a-lawnmower-telephone-attack-today/ Citibankhas been under a telephone calling attack in 20th september. Does anyone in asterisk community got any CDRs or logging of similar attacks as the one above mentioned ? Any one with logging of it or future information about the case ? Identified more detaills in this attack ? Citibank is or has been under a telephone calling attack latest 12 hours. Here I will explain the attack and how it was done. Have you seen the movie “lawnmower man”, when in the end, all phones rings in the who city? This was the aim for todays attack on Citibank in UK. The attack was simple, but probably effective when it was active. Send SIP INVITE to open SIP gateways and PBXs, who then will actually use the traditional phonesystem (POTS) to call the target. Suddenly you need DoS protection on your traditional POTS lines…. The SIP INVITE looks like this. INVITE sip:00442075005...@x SIP/2.0 Via: SIP/2.0/UDP 217.23.7.47:58585;branch=z9hG4bKaergjerugroijrgrg To: sip:x From: sip:217.23.7.47:58585;tag=Zerogij34 Call-ID: 213948958-34384780214-384...@217.23.7.47 CSeq: 1 INVITE Max-Forwards: 69 Contact: sip:s...@217.23.7.47:58585;transport=udp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,NOTIFY,REFER,MESSAGE Content-Type: application/sdp Content-Length: 520 Session-Expires: 3600; Allow-Events: refer.. v=0 o=sip 2147483647 1 IN IP4 1.1.1.1 s=sip c=IN IP4 1.1.1.1 t=0 0 m=audio 29784 RTP/AVP 8 0 4 18 18 18 18 96 3 98 a=rtpmap:96 telephone-event/8000 a=sendrecva=ptime:20 a=rtpmap:18 G729AB/8000 a=rtpmap:18 G729B/8000 a=rtpmap:18 G729A/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723 Lets walk through the SIP packet and see what info we can get from it: A quick google search on the tag: Zerogij34 reveals that this attack has been around since at least 6th of August. The IP (217.23.7.47)from this packet should be located in Portugal but the other attacks originate from both UK and Netherlands. There is no User-Agent listed, so the packet is very likely crafted from toosl like sipsak or sipp. The codec list seems real, but they use an obscure address (1.1.1.1) for the RTP. If they would use their own IP address, it could case a small DoS with RTP traffic for every successful call.)The port 29784 is within the range of Cisco units (26 000-32 000) The other INVITES reveals that the attacker is trying to figure the extension to get a dial-tone: * INVITE sip:00442075005...@67.170.104.216 SIP/2.0 * INVITE sip:011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * INVITE sip:0011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:900442075005...@67.170.104.216 SIP/2.0 * INVITE sip:9011442075005...@67.170.104.216 SIP/2.0 * INVITE sip:90442075005...@67.170.104.216 SIP/2.0 * INVITE sip:442075005...@67.170.104.216 SIP/2.0 * and several more… But is this a DoS attack on Citibank? I doubt it. Why call the Citibank on a Sunday 5 a.m.? This is more likely that Citibank has lots of lines and therefore the SIP INVITES does not generate an error (busy or others). The attacker does not hear any ringtone, but he/she should see the 180 Ringing / 180 Session in Progress. Then he or she knows that he could actually get through to the PSTN on this SIP proxy. If it would be a ringing attack, why does the attacker just send one single SIP INVITE through each gateway that actually calls this destination? The machines with the attacking IP addresses should be put under surveillance to see who connects to these. They are probably just some bots in a larger network, but they need to relay back which gateways actually responded successfully. Sad to say, but I believe this is only the small beginning…. Looking forward to hearing from you guys ;) Cheers, -- Marco Mouta ___ -- 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] Wifi GSM handover
Hello guys, I'm wondering what is required and involved in order to provide a wifi/GSM handover to customers. After googling I haven't found any product/vendor. Do you have an idea ? Thanks in advance Patrick ___ -- 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] Incoming extension not working.
I don't know if maybe you just sanitized your message for posting to the list but the number coming in from vitelity is different from what you've got in extensions.conf…also not seeing any of the necessary peer definitions in your iax.conf sample to be able to accept the call from vitelity. Thanks, --Warren Selby On Oct 9, 2009, at 4:48 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all. I'm probably doing Something Dumb(tm), so please feel free to point out whatever I'm missing, no matter how stupid. Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get: Rejected connect attempt from 64.2.142.19, who was trying to reach '6031234567@' This leads me to my first question -- why doesn't it show a context? (My second is, what's wrong with the snippets, below?): iax.conf: [vitelity] context=vitelity register = username:passw...@inbound6.vitelity.net extensions.conf: [vitelity] ; Figured I'd try both things usually used to answer... exten = 6034713217,1,Answer exten = s,1,Answer [...] [default] include = vitelity Thanks... -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- 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
Re: [asterisk-users] Wifi GSM handover
On Sat, Oct 10, 2009 at 03:15:20AM +0200, Patrick wrote: Hello guys, I'm wondering what is required and involved in order to provide a wifi/GSM handover to customers. After googling I haven't found any product/vendor. Do you have an idea ? That's called UMA and you need operator cooperation. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net ___ -- 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] Today's problem: Inbound call routing
Usually that message comes up because the caller is anonymous and freepbx doesn't like anonymous calls by default. There is an option to accept anonymous calls, or set the incoming trunk to accept calls from the specific IP address Of course it could be something else Cheers Duncan Ben Schorr wrote: Sorry, I'm brand new at Asterisk (and/or FreePBX). I'm going to have to figure out what all those things are before I can show them. I'll have to get back to you. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, October 09, 2009 9:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Today's problem: Inbound call routing On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote: -Original Message- Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 That is a pretty clear error message. Yes, I thought so. But how do I fix it? So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? You manage your dialplan with FreePBX. This mailing list supports Asterisk. I have no problem with questions about FreePBX systems. But they should also be phrased as Asterisk questions. This is a FreePBX question. I see, so this isn't an Asterisk problem it's a FreePBX problem? Creating an inbound route is FreePBX speak. This is a FreePBX question. Please ask an Asterisk question. For instance, show a dialplan trace, show the respective dialplan, show the respective channel configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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
Re: [asterisk-users] Wifi GSM handover
There are two commercial vendors that come to mind, namely DiVitas and Agito. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Sent: Friday, October 09, 2009 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Wifi GSM handover Hello guys, I'm wondering what is required and involved in order to provide a wifi/GSM handover to customers. After googling I haven't found any product/vendor. Do you have an idea ? Thanks in advance Patrick ___ -- 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
