Re: [asterisk-users] NAT issue with Fortinet Firewall
Fortinets have a SIP session-helper. Sometime this causes issues, try turning it off. To do this you need to enable telnet on the forinet management interface. Telnet into the cli and type the following config system session-helper edit 12 set port 5066 end Instead of turning this off or taking it out I am changing the port so it will not affect 5060 anymore. This way you can put it back if this doesn't work for you. John Bittner Simlab.net -Original Message- I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter localnet defines the local network and that address falls in that range, how will Asterisk treat the endpoints? I have nat=yes for all phones and canreinvite=no as well. The externip parameter is set to the outside public IP address. Still we have calls with one way audio. This is the first setup with a firewall that rewrites the IP address of the endpoint so I do not know how that is affecting the packet flow. On my other servers I can always see the public IP of the endpoint. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nortel 1140E
Retail for these phones are $300 but I think in quantity they will be $200 John Bittner Simlab.net -Original Message- You're right, these phones look great ! Do you have an idea about its price (for a 100 quantity) ? It would be interesting to know how to localize its menu ? (you couldn't localize Cisco phones with SIP firmware and Asterisk). Regards Spam Not spam Forget previous vote ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nortel 1140E
Anyone get the Nortel 1140E phones working with Asterisk ? These look like great phones and I would like to start using them on our deployments. I know these will work with Asterisk but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP please email me off list. Any help on this is appreciated. Thanks John Bittner Simlab.net 9734333011 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Cisco calling Name
The fix for this is not to use the normal Cisco IOS. Must use 12.4T version. It is a Cisco bug. John Bittner Simlab.net -Original Message- On Sat, 1 Dec 2007 00:42:43 -0500, John Bittner wrote: Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling are you sure you've got ulaw enabled for that peer in sip.conf ? and the invite trace shows that the cisco is not sending any cname. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach CanIt if this mail (ID 11581481) is spam: Spam:https://mx1.simlab.net/b.php?i=11581481m=db075974ace6c=s Not spam:https://mx1.simlab.net/b.php?i=11581481m=db075974ace6c=n Forget vote: https://mx1.simlab.net/b.php?i=11581481m=db075974ace6c=f -- END-ANTISPAM-VOTING-LINKS ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Cisco calling Name
Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI -- SIP read from 216.86.35.24:63549: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag=tgrp:PRI-TRUNK-GROUP1;branch=z9hG4bK111A56 From: sip:[EMAIL PROTECTED];tag=4F9EF08-163B To: sip:[EMAIL PROTECTED] Date: Sat, 01 Dec 2007 05:23:25 GMT Call-ID: [EMAIL PROTECTED] Supported: 100rel,timer,replaces Min-SE: 1800 Cisco-Guid: 1613584196-2667844060-2152857615-892193345 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: pending sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1196486605 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event MIME-Version: 1.0 Content-Type: multipart/mixed;boundary=uniqueBoundary Content-Length: 680 --uniqueBoundary Content-Type: application/sdp v=0 o=CiscoSystemsSIP-GW-UserAgent 6852 2375 IN IP4 216.86.35.24 s=SIP Call c=IN IP4 216.86.35.24 t=0 0 m=audio 18472 RTP/AVP 0 101 c=IN IP4 216.86.35.24 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 --uniqueBoundary Content-Type: application/gtd Content-Disposition: signal;handling=optional IAM, PRN,isdn*,,NI***, USI,rate,c,s,c,1 USI,lay1,ulaw TMR,00 CPN,04,,1,9734333001 CGN,04,,1,y,4,9733901090 CPC,09 FCI,,,y, GCI,602d57449f0411dc8052000f352dca41 UFC,GEN,5,gentf,79 UFC,GEN,5,fachd,9f8b0100 UFC,GEN,5,inpdu,02010106072a8648ce150004 --uniqueBoundary-- --- (21 headers 33 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 216.86.35.24 : 5060 (non-NAT) Found peer '216.86.35.24' Transmitting (no NAT) to 216.86.35.24:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag=tgrp:PRI-TRUNK-GROUP1;branch=z9hG4bK111A56;received=216.86.35.24 From: sip:[EMAIL PROTECTED];tag=4F9EF08-163B To: sip:[EMAIL PROTECTED];tag=as39c359be Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: SimlabVOIP Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' voippbx01*CLI -- SIP read from 216.86.35.24:5060: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 216.86.35.24:5060;x-route-tag=tgrp:PRI-TRUNK-GROUP1;branch=z9hG4bK111A56 From: sip:[EMAIL PROTECTED];tag=4F9EF08-163B To: sip:[EMAIL PROTECTED];tag=as39c359be Date: Sat, 01 Dec 2007 05:23:25 GMT Call-ID: [EMAIL PROTECTED] Max-Forwards: 70 CSeq: 101 ACK Content-Length: 0 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP multicast support
Anyone know what SIP phones support RTP multicast intercom or MOH. I am working on a project that a client needs to page 150 phones at the same time. I have clients that have 40 phones working with a custom script that I wrote that checks to see if there on the phone and if not puts them in a meetme room, but this is to slow. John Bittner Simlab.net ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI progress codes.
Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a fast busy. I am working with the carrier to get this fixed but its not going easy. Is there anyway when asterisk sees the progress code to cancel the dial and playback a message mapped to the progress code type. Any help on this would be appreciated. John Bittner Simlab.net 9734333011 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme
Anyone notice on latest SVN trunk that meetme no longer asks for a pin when you try to enter a room? I want to verify it before I post it as a bug. John Bittner Simlab.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Callerid Name
Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 572/0x23C) (Terminator) Message type: ALERTING (1) [1e 02 81 88]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- SIP/69.60.198.130-5119 is ringing Protocol Discriminator: Q.931 (8) len=36 Call Ref: len= 2 (reference 572/0x23C) (Originator) Message type: FACILITY (98) [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 42 55 4c 4b 41 4e 27 53 2c 48 45 41 4c 54 48] Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'BULKAN', 0x27, 'S', 0x2c, 'HEALTH' ] -- Processing IE 28 (cs0, Facility) Handle Q.932 ROSE Invoke component ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for Full Time technicians
We are currently looking for knowledgeable Asterisk system technicians in the NJ area. Candidates MUST be competent, qualified, efficient, expedient, trustworthy, and reliable. Must have deployed a few systems and be very familiar with all aspects of installing and configuring asterisk. The technician must be able to follow instructions as well as, work independently on service calls, installations, or as a member of a project team. **System tech's must be able to demonstrate basic trouble shooting skills!** Responsibilities: *Cost effectiveness and time efficient on the job site. *Conduct daily business in a professional and courteous manner with out disrupting the clients daily business operations. *Constantly act as a team player with the companies best interest in mind. *If presented with a challenge and unable to solve independently, seek assistance from supervisor or company President. *Maintain reliable personal transportation *Provide and maintain personal tools. Benefits: *40 hour work week *Full medical and dental coverage after 90 days *Paid Holidays *Paid vacation after six months *Company fuel card *Company Van *Company supplied logo shirts ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Most Stable Version of Asterisk
Anyone know what version of Asterisk is the most stable running Real-time queues and agents ? I am setting up a 200 phone call center and the first test run caused the system to crash 3 time in 3 days with only about 100 calls an hour. I used the same build that I have used in prior stable installations the only difference is I was not running real-time. Any help would be appreciated. Thanks John Bittner Simlab.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceMail
In your voicemail.conf file you should see global options. Uncomment out envelope=no and nextaftercmd=yes This should help http://www.voip-info.org/tiki-index.php?page=Asterisk+config +voicemail.conf John B -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Monday, June 20, 2005 9:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] VoiceMail I installed Asterisk Voicemail in an office and now most of the employees are complaining that when they're listening to the messages, it takes forever to listen to their messages. The reason being is that before the message is played, the voicemail says the full date and time when the message arrived and that takes a long time. It's like: Friday . June20th. 2000...and...5... etc (you get the idea). Is there anyway to shorten that or even give users the option to not play that? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VOIP-INFO down?
To whoever owns this site. To help keep this up and running I am willing to host it for free. I run a regional ISP in the northeast. Please contact me off list. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 14, 2005 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] VOIP-INFO down? Second day in a row... -Original Message-MYDYNDNS.ORG From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 8:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi all, Is VOIP-info down? Marcel van Kaam Fonetica Teleservices ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI over SIP
Anyone know how to get this to work? Does Asterisk support it. If not I am willing to post a bounty to get it to work. Anyone interested please email me direct. Thanks John Bittner Simlab.net 973-239-8548 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with New SIP phone.
Hi, I have been testing out a new voip phone from Comdial called EP300. I have it working with Asterisk with very good results. This phone is the best phone I have tested with a price under $150. The only issue with this phone is programming the buttons. I ran a ethereal trace on it with a comdial system and found that it uses subscribe/notifies for configuration and button status. I have attached some of the traces. I am looking for someone to work with me to get the buttons to work. I am willing to fund this project and post it back to community. Anyone interested please email me. Thanks John Bittner Simlab.net 973-239-8548 ext 229 or 1299 comdial2 Description: Binary data ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk ADSI
Hi, Anyone know if you can use a ADSI phone with Asterisk behind an Linksys ATA. I know packet 8 uses these phones with an ATA. I tried testing it but when you try to program the phone you get a error ADSI Unavailable on CPE. Do I have to program the phone with a zaptel card first before using it with an ATA? I looked all over the net for some info but not much on ADSI. Any help is appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for tech in San Francisco
Looking for a tech that can go onsite to Two Embarcadero Center, Suite 670 San Francisco, Ca 94111 to work on a simple voip networking issue. Anyone interested please call me at 9734333001 ext 226 asap. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk@home 0.6
I started working on testing [EMAIL PROTECTED] I have setup the system with 5 phones and 1 pots line. I am using polycom phones for this system. Polycom's register and can make outbound calls with no issues. When I make an internal call... The calls go straight to vm without ringing any phones. Incomming pots call do the same thing. Went crazy thinking it was a polycom config issue...but its not.. If I rem out the dial macro in exten-vm and replace it with a normal Dial(SIP/2000) the phones rings, everything works. It looks like the problem is with the dialparties.agi script. I see it exiting with 0 before going into voicemail. Anyone have any idea why? Any help would be much appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue and Normal Transfer.
Hi, Does anyone know how to get the normal transfer button to work when transferring a queue call. There seems to be a bug in app_queue that prevents calls from reaching agents. If a call is directed to an agent, and that agent transfers the call using the transfer facility on Cisco phones the call is disconnected. If the agent uses # transfers it works but the agents do not want to do blind transfers. Sometimes they also forget to use # and hang-up on calls. Is there anyway to fix app_queue to get the normal transfer buttons working. I am will to pay for this fix. Let me know Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Shared Call Appearance
Has anyone got Polycom Shared Call Appearance working with Asterisk ? If Asterisk doesn't support this, I am willing to put up a bounty of 1000 to get it to work. John Bittner Simlab.net Shared Call Appearance Signaling A shared line is an address of record managed by a server. The server allows multiple endpoints to register locations against the address of record. SoundPointR IP supports shared call appearances (SCA) using the SUBSCRIBENOTIFY method in the SIP Specific Event Notification framework (RFC 3265). The events used are: . call-info for call appearance state notification. line-seize for the phone to ask to seize the line ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue_log
Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ALERT_INFO issue CVS-HEAD-12/24/04
Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net This is a what I have in my dialplan. exten = 207,1,SetVar(ALERT_INFO=Ring Answer) exten = 207,2,Dial(SIP/207) exten = 207,3,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for Full or Part time asterisk techs
We are currently looking for knowledgeable Asterisk system technicians in the NJ area. Candidates MUST be competent, qualified, and reliable. Must have deployed a few systems and be very familiar with all aspects of installing and configuring asterisk. The technician must be able to follow instructions as well as, work independently on service calls, installations, or as a member of a project team. **System tech's must be able to demonstrate basic trouble shooting skills!** **Linux programming a major + Responsibilities: *Cost effectiveness and time efficient on the job site. *Conduct daily business in a professional and courteous manner with out disrupting the clients daily business operations. *Constantly act as a team player with the companies best interest in mind. *If presented with a challenge and unable to solve independently, seek assistance from supervisor or company President. *Provide and maintain personal tools. *Maintenance and accountability for all capital tools placed in the technicians possession by the company. Benefits: *40 hour work week *Full medical and dental coverage after 90 days *Paid Holidays *Paid vacation after six months *Company fuel card *Company Van *Company supplied logo shirts John Bittner Please email me at [EMAIL PROTECTED] with responses. Thank You. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Install Xc-Ast $$$
I have spent the last 3 days trying to get this software working. I am now at the point I am willing to pay to get this installed. Anyone that has installed this before and is looking for some cash please email me with price. I need this installed asap. Thanks John Bittner Simlab.net 9734333009 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanSpy
Anyone know why chanspy was not included in asterisk distribution as of October. ? I tried patching my current 1.0 but seems the patches are for an older version. I posted a bounty of $250 to get this to work with the newest stable. Needs be able to monitor bridged sip calls with or without a monitoring beep. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return codes from extension.conf
Anyone know how return codes work and how I can use them in
my dialplan?
I am trying to get my system to monitor how many agents are
logged into a queue. When the queue is empty the system will
forward the call to an outside number.
I tried setting a globalvar to the total number of agents
logged in and route the call based on the value of the
variable.
The problem I am having is the globalvar gets reset to 0
after about 5 hours for some unknown reason.
What I want to do is put a gotoif in my extensions.conf and
route the call based on the value of the return code.
Help!!
exten = 91,1,AddQueueMember(ultramar1)
exten = 91,2,SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} +
1])
exten = 91,3,Playback(agent-loginok)
exten = 91,4,Hangup
exten = 91,103,Congestion
exten = 91,104,Hangup
;
exten = 92,1,RemoveQueueMember(ultramar1)
exten = 92,2,SetGlobalVar(AGENTS_AVAIL=$[${AGENTS_AVAIL} -
1])
exten = 92,3,Playback(agent-loggedoff)
exten = 92,4,Hangup
exten = 92,102,Congestion
exten = 92,103,Hangup
John Bittner
Simlab.net
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[Asterisk-Users] Asterisk queue
Anyone know how to use this feature. 'leavewhenempty = yes' I got it to leave the queue if no one is logged in, but I expected it to go to priority n+101 I need it to work like this. exten = s,1,Answer exten = s,2,SetMusicOnHold(random) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,15 ;SAI menu - * ultramaradmin, # for voicemail exten = s,5,Background(ultramarmainmenu) exten = s,6,Wait,3 exten = s,7,Playback(ultramarbeforequeue) exten = s,8,Queue(ultramar1|n) exten = s,9,DigitTimeout,5 exten = s,10,ResponseTimeout,20 ;SAI menu - 1 for voicemail1, 2 for voicemail2, 3 for voicemail3 exten = s,11,Background(ultramarvoicemail) exten = s,12,Wait,5 exten = s,13,Goto(ultramarsfo,s,8) exten = s,109,Dial(SIP/[EMAIL PROTECTED]) exten = s,110,Hangup It just keeps going back into the queue. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk queue
Sorry about that ... Anyone know how to use this feature. 'leavewhenempty = yes' I got it to leave the queue if no one is logged in, but I expected it to go to priority n+101 I need it to work like this. exten = s,1,Answer exten = s,2,SetMusicOnHold(random) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,15 ;SAI menu - * ultramaradmin, # for voicemail exten = s,5,Background(ultramarmainmenu) exten = s,6,Wait,3 exten = s,7,Playback(ultramarbeforequeue) exten = s,8,Queue(ultramar1|n) exten = s,9,DigitTimeout,5 exten = s,10,ResponseTimeout,20 ;SAI menu - 1 for voicemail1, 2 for voicemail2, 3 for voicemail3 exten = s,11,Background(ultramarvoicemail) exten = s,12,Wait,5 exten = s,13,Goto(ultramarsfo,s,8) exten = s,109,Dial(SIP/[EMAIL PROTECTED]) exten = s,110,Hangup It just keeps going back into the queue. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk status retriever script
Hi,
Anyone get this script to work with a cisco phone. I cant
find much info on the script. I got it to run but all I see
is a directory list with no phone status.
I took a guess on the format of the file
internal_directory.csv ...
If anyone can look at my configs and let me know what I am
doing wrong I will be forever gratefull.
Thanks
John Bittner
Simlab.net
internal_directory.csv
'Matt Home', 1219, SIP/1219,
'Matt Confrey', 219, SIP/219,
'John Bittner', 226, SIP/226,
'Rick Deluca', 210, SIP/210,
'David Hilla', 223, SIP/223,
'Bill Higgins', 241, SIP/241,
'Lee Miragliotta', 240, SIP/240,
'Miguel Lopez', 236, SIP/236,
'Dave Home', 1223, SIP/1223
#!/usr/bin/python2.3
# status.cgi - an asterisk status retriever script
# Copyright (C) 2004 C.E. Hill Co. (UK) Ltd.
([EMAIL PROTECTED])
#
# This library/program is free software; you can
redistribute it and/or
# modify it under the terms of the GNU Lesser General Public
# License as published by the Free Software Foundation;
either
# version 2.1 of the License, or (at your option) any later
version.
#
# This library is distributed in the hope that it will be
useful,
# but WITHOUT ANY WARRANTY; without even the implied
warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See
the GNU
# Lesser General Public License for more details.
#
# You should have received a copy of the GNU Lesser General
Public
# License along with this library; if not, write to the Free
Software
# Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA
02111-1307 USA
import ast, cgi, sys, csv
#import cgitb; cgitb.enable()
#dir = [['User 1', '201', 'SIP/user1', 'SIP/user2'],
# ['User A', '202', 'SIP/usera']]
def dirload():
dir = list()
f = file(internal_directory.csv, 'r')
reader = csv.reader(f)
for row in reader:
entry = list()
for i in range(0, len(row)):
entry.append(row[i])
dir.append(entry)
f.close()
return dir
# 'Channel: SIP/stan1-997f'
received call looks like this
['Event: Status',
'Channel: SIP/stan1-563c',
'CallerID: 231',
'State: Up',
'Link: Zap/1-1',
'Uniqueid: 1078506259.9',
'ActionID: 1'],
outgoing call looks like this
['Event: Status',
'Channel: SIP/stan1-1262',
'CallerID: Tristan Hill 225',
'State: Up',
'Context: stan',
'Extension: 231',
'Priority: 1',
'Link: Zap/4-1',
'Uniqueid: 1078506511.11',
'ActionID: 1'],
# from, to, direction
status
channel = 'SIP/stan1'
status[channel] = (outgoing_call, identifier)
# calls = [(True, '231'),(False, 231)]
call = (True, '231')
def dictify(event):
d = dict()
for line in event:
l = line.split(': ', 1)
if len(l) == 2:
d[l[0]] = l[1]
else:
print === %s % line
if d.has_key('Channel'):
d['Channel'] = d['Channel'][:d['Channel'].find('-')]
return d
def examine(statusList):
status = dict()
for eventList in statusList:
e = dictify(eventList)
if e.has_key('Event') and e['Event'] == 'Status':
if e.has_key('Extension'): # outgoing
status[e['Channel']] = (True,
e['Extension'])
elif e.has_key('CallerID'): # incoming
name =
ast.callerid(e['CallerID']).preferName()
status[e['Channel']] = (False, name)
return status
print Content: text/xml
print Refresh: 10
print
try:
dir = dirload()
m = ast.manager('69.60.XXX.XXX', 5038)
m.login('user', 'password')
slist = m.status()
#print slist
status = examine(slist)
m.logoff()
except:
print !-- Exception: %s, %s -- % \
(sys.exc_info()[0], sys.exc_info()[1])
dir = list()
status = dict()
print CiscoIPPhoneDirectory
print TitleInternal Numbers/Title
print Prompt%d numbers/Prompt % len(dir)
for row in dir:
name = row[0]
for channel in row[2:]:
if status.has_key(channel):
call = status[channel]
if call[0]:
name += -%s % (call[1])
else:
name += -%s % (call[1])
print DirectoryEntry
print Name%s/Name % cgi.escape(name)
print Telephone%s/Telephone % row[1]
print /Directory
print /CiscoIPPhoneDirectory
'John Home', 1229, SIP/1229
status.cgi
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[Asterisk-Users] Voicemail Macro issue.
Hi, Anyone know how to get voicemail to continue running the next exten in the dialplan when a user hangs up. If a user hits # after leaving a message instead of hanging, up it works. I am trying to do a call back macro and when users hangup after leaving a voicemail the rest of my macro does not run. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Macro issue.
Hi, Callback is what I based my script on. The problem I am having is when someone leaves a messages and then hangs up, the rest of the macro does not continue to run. If after I leave a message I hit # it works perfect. Any ideas? John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, November 08, 2004 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voicemail Macro issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Monday, November 08, 2004 10:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Voicemail Macro issue. Hi, Anyone know how to get voicemail to continue running the next exten in the dialplan when a user hangs up. If a user hits # after leaving a message instead of hanging, up it works. I am trying to do a call back macro and when users hangup after leaving a voicemail the rest of my macro does not run. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This might work for you http://www.voip-info.org/wiki-Asterisk+tips+callback It will not work for us, because we need a repeated call out until the message is picked up, so I have posted a bounty, if the feature I am looking for interests you enough to contribute please add your contribution to the bounty, at some point it will be attractive enough for a coder to do the work. Details on the bounty are here http://www.voip-info.org/tiki-index.php?page=Asterisk%20boun ty %20outcall %20notification%20application ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom failed registration - Cant figure outwhats wrong
Hi, Remove the 614p@ from reg.1.address=[EMAIL PROTECTED] John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Friday, October 29, 2004 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom failed registration - Cant figure outwhats wrong Can anyone tell me if the below is wrong for the phone configuration, it keeps failed registration. (I had this working but lost all my tftp config files so I know its a work configuration) 614p is my username password is my password and 10.20.30.3 is the asterisk box Thanks in advance. reg reg.1.displayName=614p reg.1.address=[EMAIL PROTECTED] reg.1.label=614p reg.1.type=private reg.1.thirdPartyName=614p reg.1.auth.userId=614p reg.1.auth.password=password reg.1.server.1.address=10.20.30.3 reg.1.server.1.port=5060 reg.1.server.1.transport= reg.1.server.1.expires=360 reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=0 reg.1.acd-agent-available=0 -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong
I just read what I typed... I meant to say put the 614p in the reg.1.address field with out the ip. reg.1.address=614p Sometimes I am dyslexic. John B -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, October 29, 2004 5:18 PM To: 'Matthew Marlowe'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom failed registration - Cant figureoutwhats wrong Hi, Remove the 614p@ from reg.1.address=[EMAIL PROTECTED] John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Friday, October 29, 2004 5:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom failed registration - Cant figure outwhats wrong Can anyone tell me if the below is wrong for the phone configuration, it keeps failed registration. (I had this working but lost all my tftp config files so I know its a work configuration) 614p is my username password is my password and 10.20.30.3 is the asterisk box Thanks in advance. reg reg.1.displayName=614p reg.1.address=[EMAIL PROTECTED] reg.1.label=614p reg.1.type=private reg.1.thirdPartyName=614p reg.1.auth.userId=614p reg.1.auth.password=password reg.1.server.1.address=10.20.30.3 reg.1.server.1.port=5060 reg.1.server.1.transport= reg.1.server.1.expires=360 reg.1.server.1.register= reg.1.server.1.retryTimeOut= reg.1.server.1.retryMaxCount= reg.1.server.1.expires.lineSeize= reg.1.acd-login-logout=0 reg.1.acd-agent-available=0 -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Performance (Cisco AS5350) or Price (WildcardTE410P)
In our setup we tried to use a 4 Port T1 card in a Compaq proliant. We had no issues with processor load or dropped calls but major echo problems. I figured that since I was using a PRI it was digital and should have no echo issuesI was wrong. I had just as many problems with my pri's as I had with fxo ports. I tried everything to clean it up. Went with a Cisco 5400 and haven't had any issues at all. The unit works like a champ. The unit new with 4 T1 ports is $12K ...where did you get 40K from ? Another option is to put in an echo can (DITECH) in front of the digium card. I haven't tested this yet ... John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Crocker Sent: Tuesday, October 26, 2004 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Performance (Cisco AS5350) or Price (WildcardTE410P) I'm looking to build an Asterisk system to place in front of my call center switch. My plan is to eat up 4 PRIs in one office, send to my other office and convert back to PRI for my legacy switch. Voice quality is critical, would I be better off going with a Cisco AS5350 using hardware DSPs for the g.729 codec or is the Digium g.729 codec just as good with enough CPU to throw at it. The lack of dedicated DSP concerns me Option 1 PSTN PRI - AS5350 - (2 x T1) IP(SIP,g.729) - AS5350 - PRI - PBX I would have 1 Asterisk server handling SIP, Conference, IVR, Voice Mail. I would purchase a bunch of g.729 codec licenses Option 2 PSTN PRI - Asterisk - (2x T1) IP (IAX2, g.729) - Asterisk - PRI - PBX The first Asterisk would handle SIP, Conference, IVR, Voice Mail If the voice quality of the software g.729 codec is as good as the Cisco hardware codec I would *much* rather give my money to Digium, buy their hardware and licenses. If the quality isn't as good I may be forced to spend $40k on AS5350s I'm thinking 2.8 Ghz P-IV HT with 2 GB RAM for each Asterisk server. -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Lucent Definity
Hi, I linked Asterisk to a Lucent Definity via a T1 port. I was wondering if anyone has found a way to use Lucent's ACD from the asterisk side. I looking for someone that specializes in Definity ACD and CentreVu to help me with this. I am more then willing to pay for this. Let me know Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mwi over serial port
Thats not true. I am very serious. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Wednesday, October 13, 2004 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mwi over serial port The bounty is bogus, the offerors are not serious, and they should take it off the wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Echo
Hi, Just install 6 new polycoms at a customer and all of them have a major echo issue. Have asterisk connected to the PSTN via digium 4 port fxo card in a P4 running fedora. I have tweaked zapata ran ztmonitor... just as a test I attached a cisco 7960 the cisco has no echo problems. Another issue I came up on is when I enable echo training I no longer can hear any inbound voice. Does anyone know if there is a setting in the polycom config that will cause this. The echo on the phones happen only on my side and only when I speak. The polycoms play back my voice but delayed. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF after answer
Hi, I am trying to link up a comdial PBX to Asterisk using T1 tieline EM. I have it working for comdial to asterisk but not the other way. Comdial does not listen for any DTMF before answering the ZAP channel and requires codes before allowing asterisk to call an outside line or inside extension. Does bnyone know how to get Asterisk to dial a ZAP group (g1) wait for answer then send it DTMF. I have tried the D option but I cant seem to get it work. Any help will be very appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone see this before. ?(meetme)
Hi Jul 20 16:59:12 WARNING[1242768448]: app_meetme.c:924 conf_run: Unable to writey == Spawn extension (voicepulse-outgoing, 8000, 6) exited non-zero on 'SIP/241' -- Executing Hangup(SIP/241-f931, ) in new stack I was in a conference and it just hung up on me. This is the 5 time it did this. I am running Asterisk CVS-HEAD-07/12/04-20:57:22 I was on a Polycom 500 phone. The other 2 callers in the conference did not get cut off. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soekris Engineering net4801
Hi, We used 512meg compact flash running debian. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of W. Kevin Hunt Sent: Thursday, June 17, 2004 8:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Soekris Engineering net4801 John Bittner wrote: Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. What storage medium did you use, compact flash for 2.5 HD ? What OS/flavor did you use? W. Kevin Hunt CCIE #11841 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Soekris Engineering net4801
Hi, I have it working great. I have debian running on it with music on hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with calls on all 10 phones at the same time through voicepulse with no issues. I ran top with all the phones running and I was only up to 45% cpu. Seems to run ok but I am still in the testing phase. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Senad Jordanovic Sent: Wednesday, June 16, 2004 3:26 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Soekris Engineering net4801 Has anyone tried to run * on Soekris Engineering net4801 board? If so, what were the results in terms of performance? Ta SJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoicePulse broken?
I am having an issue with voicepulse also. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C. Sullivan Sent: Thursday, May 20, 2004 12:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] VoicePulse broken? Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Immix C3-FXO gateway
Hi, Anyone get the Immix C3-FXO Sip gateway to work with asterisk. I have it working for outbound calls but cant get it to work for inbound calls. The unit has an built-in greeting and it keeps picking up the call. Cant find the command to turn it off and set it to forward the calls to asterisk. Any help on this would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI Mar 3 12:55:05 WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) This has been brought up in the previous post but it does not seem to have an answer for it so far. I cvs the stable v1.0 this morning after compiling and installing I have calls drop 1 minutes into the connection with the above message. If anyone has any idea of this occurrence. I have set up sip.conf: canreinvite=no -- David Kwok Tel: 612 99292086 ext 1002 Iaxtel/FWD # 17001813482 ext 1002 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SCO finds someone to pay!!!
Does anyone know if they took out SCO's code in Linux 2.6 kernel ? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex LopezSent: Tuesday, March 02, 2004 11:31 PMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] SCO finds someone to pay!!! I dont believe this!! SCO got some one to pony up 7 figures!! http://www.eweek.com/article2/0,1759,1541140,00.asp
RE: [Asterisk-Users] Tiny install with Solid State Storage
I have a unit running Redhat 9 on a 1 gig flash card. Since a 1 gig flash card is expensive I am working on a unit running http://www.trustix.net/ on a 256meg flash card. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, March 01, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Tiny install with Solid State Storage Hello All, I was wondering if anyone is successfully running asterisk on a system with solid state storage, such as a compact flash card? I'm looking for some pointers on doing this. Thanks -Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues
Hi, Does anyone know how to check the status of a queue from within extensions.conf. If a queue has no one logged into it I want to redirect the call to a manager phone. Any ideas would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Failed to start asterisk
Had the same issue. My Mini-itx board has a VIA C3 processor and this fixed
this issue for me.
Set PROC in the main Makefile of asterisk to i586 then recompile.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of dkwok
Sent: Friday, February 27, 2004 5:28 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Failed to start asterisk
I am using mini-itx motherboard and I installed asterisk stable from
cvs. However below is the messages when starting asterisk by
safe_asterisk. Anyone spotted the cause of not starting.
Last login: Fri Feb 27 10:40:44 2004
[EMAIL PROTECTED] root]# safe_asterisk
[EMAIL PROTECTED] root]# /usr/sbin/safe_asterisk: line 77: 3448 Illegal
instruction (core dumped) asterisk ${ASTARGS} 1/dev/${TTY}
/dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3463 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3478 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 77: 3493 Illegal instruction (core
dumped) asterisk ${ASTARGS} 1/dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 132
Asterisk exited on signal 4.
Automatically restarting Asterisk.
/usr/sbin/safe_asterisk: line 44: /dev/tty9: Input/output error
/usr/sbin/safe_asterisk: line 45: /dev/tty9: Input/output error
Asterisk ended with exit status 1
Asterisk died with code 1. Aborting.
--
David Kwok
Iaxtel/FWD # 17001813482 ext 1002
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[Asterisk-Users] ACD
Hi, I saw on the bugs.digium.com site a feature patch that does the following. Announce callers position in queue every x seconds / service level reporting / queue timeouts / hold time estimation It looks like it was writen for an older version of asterisk. I am looking to get this to work on CVS version 01/30/04 I am willing to pay for this. Let me know Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones
Hi, Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. Any ideas would be helpfull. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones
Hey Dave, I tried that.. no change. Keep in mind that a regular pots phone works ok. Only having this issue with aastra PT390 phones. Is there something I am missing. Is the signaling different with ADSI phones. I have ADSI on in the zapata.conf I plug the PT390s into a normal pots line and they work. Anyone ever get these phone working with asterisk. Any help would be appreciated. Thanks John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Weis Sent: Saturday, February 14, 2004 9:33 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones On Fri, 13 Feb 2004, John Bittner wrote: Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. I had a similar problem with an adtran TA750 with digits not breaking dialtone. It would come and go, usually working fine right after a restart. dave -- Dave Weis I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations.- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Rhino channel bank and aastra PT390 phones
Hi, Anyone setup a Rhino channel bank ?... any issues. I got it working with normal pots phones but I cant get it to work with Aastra PT390 phones. The phones get dialtone but the asterisk does see any DTMF digits dialed from the phone. Any ideas would be helpfull. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialout redunancy.
Hi, How do I get asterisk to use an alternate outbound provider in the event my primary IAX provider goes down. I currently have an IAX provider that is having issues, so I signed up with a sip provider for a backup. I added the sip provider info into the extensions.conf file as the second outbound entry, but asterisk still tries to call the iax provider 1stand since the call is incomplete the end-user hangs up.Any ideas would behelpful. Thanks John Bittner Simlab.net
RE: [Asterisk-Users] dialout redunancy.
I got it working by configuring qualify in my iax.conf. I guess asterisk
didn't think the IAX provider was down until I added that line.
As for incoming I have an 800 number pointing to 2 local phone numbers. 1 on
voicepulse and 1 on voiceglo. This way if voicepulse is down it will route
the call to voiceglo. Hopefully as the voip providers get better they will
offer a forwarding feature. Vonage does.
John Bittner
Simlab.net
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Matthew B Marlowe
Sent: Sunday, February 08, 2004 5:45 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redundancy.
Dialout redundancy using this method works perfect. I've
been using this method for some time now. I currently have
two IAX2 providers and plan to get another backup as well (In
addition to me getting my Digium cards tomorrow that'll be
another backup.)
That's great for outgoing calls, but... I'm trying to figure
out the best approach to use for incoming calls.
I currently have a VP phone number, it's the only incoming
number I have for the other voip providers I have don't offer
local termination (or any at all for that matter).
We have a POTS line from Verizon and we'd like to continue
using that phone number.
Originally we were just going to forward that phone number to
VP. But what happens if VP goes down? I figure in that case
(and we'd have to get in touch with VP if they will forward
to another number if they're done), to then forward to
another voip / pots line that we have.
Is there any other approach we can use to do this?
Possibly, a service that'll offer something like:
Transfer to 1609xxx but if busy, forward to 1609xxx,
etc. and so on?
In addition does anyone know where I might be able to port my
number to that supports transferring instead of forwarding?
I currently have Verizon and they said we need a CustoFlex
plan which will only support 6 forwards so if 7 callers
call in, the 7th will get a busy signal.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Brent Franks
Sent: Sunday, February 08, 2004 3:15 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] dialout redunancy.
You will need to set priorities for each one.
For example:
exten = _91NXXNXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91NXXNXX,2,Playback(pstnallbusy)
exten =
_91NXXNXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
exten = _91NXXNXX,4,Congestion
Basically what happens here, is I try to put it out on the
Verizon POTS
lines first, then if that doesn't work, I play a message
saying all the
lines are busy, hold if the call is important (it's now billable), the
user holds, and it goes to voicepulse.
You could get rid of the All Busy message if you wanted, I
just like to
know that the call is going to be billed (since I have unlimited LD on
my POTS lines). If that fails, It plays a fast busy.
You can also do a qualify in your iax.conf and sip entries to know
whether they are reachable before trying the call. Read up on
qualify to
find out how to do it for your needs.
Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
John Bittner
Sent: Sunday, February 08, 2004 2:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialout redunancy.
Hi,
How do I get asterisk to use an alternate outbound provider
in the event
my primary IAX provider goes down. I currently have an IAX
provider that
is having issues, so I signed up with a sip provider for a backup. I
added the sip provider info into the extensions.conf file as
the second
outbound entry, but asterisk still tries to call the iax provider
1st and since the call is incomplete the end-user hangs up. Any ideas
would be helpful.
Thanks
John Bittner
Simlab.net
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[Asterisk-Users] RE:voiceglo sip config
Hi, I am trying to get voiceglo to work with asterisk. I have tried many sip configs and cant seem to get it to register. Please if someone can look at this softphone config and let me know what I am doing wrong I would appreciated it. Thanks John Bittner Simlab.net This is my config and the softphone config listed below. [general] port=5060 bindaddr=0.0.0.0 tos=lowdelay disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED] [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=myphone.voiceglo.com context=incoming [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] RedirectAutoIgnore=dword: UseApplicationSIP=dword: RedirectIgnore=dword: SeparateRegistrarAddress=172.19.1.88 UseOutboundProxy=dword:0001 SendINVITEWithoutOffer=dword: FWDBehindNAT=dword: ReRegistrationInterval=dword:0e10 RedirectDND=dword: UseSeparateRegistrarAddress=dword: RegisterOnProxy=dword:0001 QuietlyAcceptRedirect=dword:0001 RestrictCallerIdentity=dword: DisableNonProxiedCalls=dword: IgnoreRefer=dword: ConfirmTransferRequests=dword: ExposeSoftwareVersion=dword:0001 UnregisterContactOnly=dword:0001 ProxyPort=dword:13c4 TrafficDumpFileName=C:\\SIPTRAFFIC.LOG CompatibilityFlag1=dword: TrafficDumpRingBufferLength=dword:00ff SeparateRegistrarPort=dword:13c4 PreferredRegistrationTCP=dword: WorkThroughProxyOnly=dword: ProxyAddress=myphone.voiceglo.com AddressOfRecord=sip:973111.voiceglo.com ProxyUserName=973111 ProxyUserPassword=UPUIOPHXDTV ProxyDomain=myphone.voiceglo.com CallerNumber=973111 RedirectionURL= FWDNumber= FWDPassword= SeparateRegistrar= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE:voiceglo sip config
Hi, After allot of trial and error I found what I did wrong. I was missing the port. This config works if anyone needs it. Voiceglo config [general] port=5060 bindaddr=0.0.0.0 disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED]:5060 [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=dynamic nat=yes port=5060 context=incoming John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Friday, February 06, 2004 6:07 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RE:voiceglo sip config Hi, I am trying to get voiceglo to work with asterisk. I have tried many sip configs and cant seem to get it to register. Please if someone can look at this softphone config and let me know what I am doing wrong I would appreciated it. Thanks John Bittner Simlab.net This is my config and the softphone config listed below. [general] port=5060 bindaddr=0.0.0.0 tos=lowdelay disallow=all allow=gsm allow=ulaw allow=alaw maxexpirey=180 defaultexpirey=160 tos=reliability register=973111:[EMAIL PROTECTED] [myphone.voiceglo.com] type=friend secret=UPUIOPHXDTV username=973111 host=myphone.voiceglo.com context=incoming [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] [HKEY_CURRENT_USER\Software\SJLabs\SJvoip Project\SJphone\Options\SIP 2.0] RedirectAutoIgnore=dword: UseApplicationSIP=dword: RedirectIgnore=dword: SeparateRegistrarAddress=172.19.1.88 UseOutboundProxy=dword:0001 SendINVITEWithoutOffer=dword: FWDBehindNAT=dword: ReRegistrationInterval=dword:0e10 RedirectDND=dword: UseSeparateRegistrarAddress=dword: RegisterOnProxy=dword:0001 QuietlyAcceptRedirect=dword:0001 RestrictCallerIdentity=dword: DisableNonProxiedCalls=dword: IgnoreRefer=dword: ConfirmTransferRequests=dword: ExposeSoftwareVersion=dword:0001 UnregisterContactOnly=dword:0001 ProxyPort=dword:13c4 TrafficDumpFileName=C:\\SIPTRAFFIC.LOG CompatibilityFlag1=dword: TrafficDumpRingBufferLength=dword:00ff SeparateRegistrarPort=dword:13c4 PreferredRegistrationTCP=dword: WorkThroughProxyOnly=dword: ProxyAddress=myphone.voiceglo.com AddressOfRecord=sip:973111.voiceglo.com ProxyUserName=973111 ProxyUserPassword=UPUIOPHXDTV ProxyDomain=myphone.voiceglo.com CallerNumber=973111 RedirectionURL= FWDNumber= FWDPassword= SeparateRegistrar= ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dialing delay question.
Hello. I have been working on getting my asterisk box to connect to a lucent definity PBX using a T100p. I connected it to a t1 port on the lucent box and ran a T1 crossover cable to the t100p. I have asterisk setup with a voicepulse account for outbound and inbound dialing. On the lucent pbx I matched the configs of a working channelized MCI t1. The only thing asterisk is doing is acting as a voip gateway. Everything works ok with only one issue. When someone makes a call outbound and hesitates on dialing the numbers, asterisk only picks up a partial number. It doesn't wait until the person dialing finished dialing the full number. If the person dials the number without any delay the call goes through. On the asterisk console I see the partial phone numbers when this happens. How does this type of setup work. Is this a lucent issue or an asterisk issue. Does the lucent pbx just open a channel, send digits and them the asterisk box has dialplan time out or is the lucent pbx to wait until it sees all the numbers before sending it to the asterisk box. Any help will be appreciated. Thanks John Bittner Zapata.conf [channels] ; ; T100P plugged into Lucent Definity ; context=voicepulse-outgoing signalling=em_w group=1 channel =1-24 rxwink=300 usecallerid=yes callwaiting=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callerid=asreceived ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Appliance
We have done this with nexcom.com appliances. We are currently using EBS 1569PS unit with a 1gig flash card. Keep in mind were only using the gateway features of asterisk on this box. If you are using the full PBX features you may want to use a more powerful model. John Bittner Simlab.net Nexcom contact info. Brian D. Earnhart Regional Sales Manager Nex Computer Inc. 46707 Fremont Blvd. Fremont CA 94538 510-656-2248 x 13 510-656-2158 Fax 510-396-7753 Mobile [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Martin Sent: Wednesday, January 28, 2004 12:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Appliance Ok now that we have a Asterisk server running quite well, we want to put it onto a more appropriate device, i.e. not a beige box computer, but perhaps some kind of embedded linux appliance. Has anyone already done this? Any suggestions on some tidy, small, suitable linux systems to use for asterisk? I.e., somthing that looks like this: https://secure.makonetworks.com//images/main/mako_250_shad.gif ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
