[Asterisk-Users] call routing based on dnis
Hi, Is it to possible to route incomming call using dnis information to specific extension or section in the extensions.conf ?? like my users dial my asterisk box using different toll free numbers, but i can't offer them unique services. Plz suggest that if it is possible ? if possible then any example ... Regards Apna Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.
Re: [Asterisk-Users] call routing based on dnis
azher: simple to do, assuming numbers are being passed through on dnis, in the relevant context (from zapata) put exten = 6601122,1,Hangup #users dialing here bye-bye exten = 5551122,1,Playback(beep) #users here (pun intended) beep - wasim On Sun, 17 Aug 2003, Azher Amin wrote: Is it to possible to route incomming call using dnis information to specific extension or section in the extensions.conf ?? like my users dial my asterisk box using different toll free numbers, but i can't offer them unique services. Plz suggest that if it is possible ? if possible then any example ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail2 patches
Yeah, I haven't really had time the last couple of weeks to follow up on this. I'd be happy to have any of the patch in CVS if that were so desired, but before heading down that road I'd be curious to know if anyone besides me has had a successful time actually using the patches. At the very least there is a bit of fine tuning for me to do in the patch, and I don't have matching prompts, just my own voice for now. Brad On August 16, 2003 02:13 pm, Brian West wrote: A few weeks ago Brad posted his patches to the mailing list: http://www.universaltime.org/~brad/vmail/ But I can't find his email address... does anyone happen to have his address. I hope he would be willing to see if Mark wanted to add those options to the CVS. I think you need to fill out a disclaimer and post it to bugs.digium.com bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream Budgetone
Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone
RRP: $75 for 101, $85 for 102 - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 1:22 PM Subject: [Asterisk-Users] Grandstream Budgetone Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_h323 one way audio
Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi compile errors with latest CVS
Title: Message Did something change in lock.h lately? I get all kind of ast_mutex errors when trying to compile chan capi 0.24c with the latest asterisk code Betel ConsultancyAbelenlaan 19 T: +31 20 640 30181185 RT Amstelveen E: [EMAIL PROTECTED]The NetherlandsW: www.betel.nl Confidentiality Notice - The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the Addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail!
Re: [Asterisk-Users] Voicemail2 patches
I haven't tried the patches, but they sounds very useful! My 2 cents... BTW, there have been some recent bug fixes to Voicemail2, so you might want to test them against recent CVS (8/16 or later) On Sunday, August 17, 2003, at 11:16 AM, Brad Bergman wrote: Yeah, I haven't really had time the last couple of weeks to follow up on this. I'd be happy to have any of the patch in CVS if that were so desired, but before heading down that road I'd be curious to know if anyone besides me has had a successful time actually using the patches. At the very least there is a bit of fine tuning for me to do in the patch, and I don't have matching prompts, just my own voice for now. Brad On August 16, 2003 02:13 pm, Brian West wrote: A few weeks ago Brad posted his patches to the mailing list: http://www.universaltime.org/~brad/vmail/ But I can't find his email address... does anyone happen to have his address. I hope he would be willing to see if Mark wanted to add those options to the CVS. I think you need to fill out a disclaimer and post it to bugs.digium.com bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi compile errors with latest CVS
ast_pthread_muxtex_* functions were changed to ast_mutex_* for the possibility of debugging the mutexes with gdb. regards Martin On Sun, 17 Aug 2003, Michiel Betel wrote: Did something change in lock.h lately? I get all kind of ast_mutex errors when trying to compile chan capi 0.24c with the latest asterisk code Betel Consultancy Abelenlaan 19 T: +31 20 640 3018 1185 RT Amstelveen E: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] The Netherlands W: http://www.betel.nl/ www.betel.nl Confidentiality Notice - The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the Addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anyone got sip/IAX working behind a firewall?
I have read loads of the emails on the subject but have not read a setup I could use. I have a windows box using Winproxy to dial up the internet through a DSL modem. My internal network I am using while developing. My internet network has been renumbered so that I could test using Nikotel4Mac. It will only accept certain numbering schemes so I have changed my ip's on the internal network to be 192.168.0.x Winproxy gets the external ip when it dials up. External (x.x.x.x) - Winproxy (192.168.0.1) - Linux (192.168.0.2) External (x.x.x.x) - Winproxy (192.168.0.1) - Windows (192.168.0.3) External (x.x.x.x) - Winproxy (192.168.0.1) - 54g radio (192.168.0.4) External (x.x.x.x) - Winproxy (192.168.0.1) - Mac OSX (192.168.0.6) Do I need to setup Winproxy so that it forwards incoming connections to asterisk on Linux? I understand that I can setup IAX to register it's connections with iaxtel.com but so far it does not make the connection. There is also lots of discussions saying that sip cannot work this way round. Is this true? I cannot put linux outside the firewall unless I change my configuration so that linux dials the net. However it is still dialling with a dynamic address so I will never know what this address is. I have also looked at the Mac OSX IAX beta soft phone called WX but that will not work either. Since I am testing using soft phones not hardware phones it all comes down to the numbering and getting the firewall to correctly send the connection requests to the right applications on the right machines. Has anyone got a similar setup working? Thanks Fats. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Configuring iptables to allow sip and dynamically allocate rtp ports
Hi, We have an asterisk box, with 2 nics, one with internal addressing, and the other with a public address. The firewall (iptables) is configured for nat routing. Now we want to allow this box to receive sip registrations from the internet. Does anyone know if you can use iptables to allow the dynamic creation of rtp ports? T
RE: [Asterisk-Users] Grandstream Budgetone
$75 for the single ethernet port version and $85 for the dual ethernet port version. You can get two for $129 at www.sipphone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Budgetone Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail2 patches
I'm sorry I'm coming in on the rear end of this but is there a difference between Voicemail and Voicemail2 in asterisk? If so what is it? AJ On Sun, 17 Aug 2003, Paul Cheng wrote: I haven't tried the patches, but they sounds very useful! My 2 cents... BTW, there have been some recent bug fixes to Voicemail2, so you might want to test them against recent CVS (8/16 or later) On Sunday, August 17, 2003, at 11:16 AM, Brad Bergman wrote: Yeah, I haven't really had time the last couple of weeks to follow up on this. I'd be happy to have any of the patch in CVS if that were so desired, but before heading down that road I'd be curious to know if anyone besides me has had a successful time actually using the patches. At the very least there is a bit of fine tuning for me to do in the patch, and I don't have matching prompts, just my own voice for now. Brad On August 16, 2003 02:13 pm, Brian West wrote: A few weeks ago Brad posted his patches to the mailing list: http://www.universaltime.org/~brad/vmail/ But I can't find his email address... does anyone happen to have his address. I hope he would be willing to see if Mark wanted to add those options to the CVS. I think you need to fill out a disclaimer and post it to bugs.digium.com bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone got sip/IAX working behind a firewall?
I use asterisk behind my Linux firewall with no problem. For IAX I have the firewall forward udp port 5036 to the asterisk box. Really simple nat setup. My asterisk box also connects to other asterisk/IAX servers through nat from behind the firewall. I'm not sure about sip at all but I know that IAX works fine. I don't know anything about Winproxy but I would assume you could easily masquerade outgoing packets/connections and then for incoming connections forward port udp 5036 to your asterisk box. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pre-newbie - some basic questions...
Hello All, Been completely obsessed for the last two days with VoIP and Asterisk - running on 2 hours sleep and coffee - sorry if this is a little scattered... Okay, I've got a small start-up company that installs traditional PBX (Nortel mainly) systems, data network infrastructure, commercial audio/video, residential audio/video/voice/data and we do lighting control systems... I've got two home offices about 30 miles apart - running some basic lan services over a CIPE link between here and there... this all started because I'd like to find a way to integrate the phone lines/system better... What really has me intriqued at the moment is the idea of using a VoIP service provider to make and recieve calls with an (800) for customers to call in on... iconnecthere.com seems to be telling me that I can do this for only $20-$25/month with no contract... Is this true/possible or am I missing something? What about nufone.net - any one have experience woth their rates and service? I'm envisioning a system in which * could handle incoming/outgoing calls to the service provider - and we could have an IP phone (maybe the Grandstream 100's for $130 a pair?) in each office and be able to answer incoming calls from either location - transfer calls between offices - access voicemail - etc... this sound familiar to anyone - any experience with this type of scenerio? inet here is a SDSL link @ 1.5megs - other office has Cable modem... I think with the encryption overhead I can only get about 14kb/sec when pulling a webpage over the VPn from an APACHE server in the other office... will I have enough bandwidth to handle calls? Thanks so much for any insights... with a new business, a 3 year-old, a 1 year old and a very pregnant wife - I don't have a lot of time to experiment with new ideas :) So I'm trying to really get a hadle on everything before I decide to invest a lot of time trying to make this work... thanks again... dave redmore ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] LAN switches with PoE? PoE phones?
Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of power hubs, but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be special for IP phones in anyway? QoS support? Managed? Also, are there PoE phones that work with *? Most I look at seem to be powered from AC wall blocks. We'd like to centralize the switching and power and provide a UPS so the phone system works when the power goes out. [Apologies, I'm new to this whole concept of IP phones and *.] -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail2 patches
On Sunday 17 August 2003 12:17, [EMAIL PROTECTED] wrote: I'm sorry I'm coming in on the rear end of this but is there a difference between Voicemail and Voicemail2 in asterisk? If so what is it? AJ All new features are going into Voicemail2. Voicemail (1) is being EOLed. Among the new features: - customizable emails - customizable 'received when' messages when checking voicemail (systemwide and per-user) - voicemail can be received in different timezones for different users - email pages when voicemail received -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone
Are these locked to the service, though? Look what vonage managed .. :) -d At 12:36 PM 8/17/2003 -0400, you wrote: $75 for the single ethernet port version and $85 for the dual ethernet port version. You can get two for $129 at www.sipphone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Budgetone Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LAN switches with PoE? PoE phones?
Hi Mike, Cisco makes PoE switches, either at the Cat 29xx or the Cat 35xx levels. The 29xx don't have gige uplinks, but the 35xx's do via GBIC interfaces. Meaning you will also need to get a GBIC media converter depending the media type (copper fiber, etc) And of course Cisco makes PoE based phones 7940 7960 which work well with * Grandstream currently requires a wall-wart, but later models are suppose to use PoE as well. I'd personally put the phones on their own subnet so that ACL filtering at the router will be easier, static IP alloc will be easier. hope this helps john brown chagres technologies, inc sip: [EMAIL PROTECTED] ptsn: (01) 505 830 1200 USA On Sun, Aug 17, 2003 at 12:44:43PM -0500, Mike Ciholas wrote: Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of power hubs, but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be special for IP phones in anyway? QoS support? Managed? Also, are there PoE phones that work with *? Most I look at seem to be powered from AC wall blocks. We'd like to centralize the switching and power and provide a UPS so the phone system works when the power goes out. [Apologies, I'm new to this whole concept of IP phones and *.] -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream Budgetone
They don't seem to be locked to the service. I've ordered several sets and changed the configs to use my AsT box with no problems On Sun, Aug 17, 2003 at 12:49:26PM -0500, denon wrote: Are these locked to the service, though? Look what vonage managed .. :) -d At 12:36 PM 8/17/2003 -0400, you wrote: $75 for the single ethernet port version and $85 for the dual ethernet port version. You can get two for $129 at www.sipphone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:23 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream Budgetone Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone
On Sun, 2003-08-17 at 19:49, denon wrote: Are these locked to the service, though? Look what vonage managed .. :) According to a thread on the ser list no they are not. But when I checked on the site for a price I found the shipping would be around 75US$ to France and then French customs would get hold of them another +20%. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LAN switches with PoE? PoE phones?
The Snom VoIP phones support PoE and Nortel makes switches: http://www.nortelnetworks.com/products/02/bstk/switches/baystack_460/ I am not certain that they are compatible, as I have not used the Snom phones and have only used the Nortel switches with PoE adapters at the other end to power wireless access points or small hubs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Ciholas Sent: Sunday, August 17, 2003 1:45 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] LAN switches with PoE? PoE phones? pamAssassin 2.55 (1.174.2.19-2003-05-19-exp) Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of power hubs, but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be special for IP phones in anyway? QoS support? Managed? Also, are there PoE phones that work with *? Most I look at seem to be powered from AC wall blocks. We'd like to centralize the switching and power and provide a UPS so the phone system works when the power goes out. [Apologies, I'm new to this whole concept of IP phones and *.] -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuring iptables to allow sip and dynamically allocate rtp ports
Not sure about making the ports dynamic but I just opened up inbound UDP ports 1-2 as in rtp.conf (these can no doubt be changes to what ever you needs require) which has been working well so far.. Hi, We have an asterisk box, with 2 nics, one with internal addressing, and the other with a public address. The firewall (iptables) is configured for nat routing. Now we want to allow this box to receive sip registrations from the internet. Does anyone know if you can use iptables to allow the dynamic creation of rtp ports? T -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LAN switches with PoE? PoE phones?
To follow up on this, the Cisco Switches and the Cisco phones will work together to create two vlans: a prioritized vlan for the phone traffic, and a secondary 10/100 link for a computer which can be attached to the phone's second switched ethernet port. Some config is needed in the switch and router to make this happen properly. I have it running well with 79x0 phones, 3550 switch, and 1751-V router. What this means is only one switch port is needed to run both a phone and a computer. This helps on the switch/phone ROI calcs. The PoE can also be used to power wireless access points. Ray Burkholder 519 570 0689 x2002 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Brown Sent: August 17, 2003 13:52 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] LAN switches with PoE? PoE phones? Hi Mike, Cisco makes PoE switches, either at the Cat 29xx or the Cat 35xx levels. The 29xx don't have gige uplinks, but the 35xx's do via GBIC interfaces. Meaning you will also need to get a GBIC media converter depending the media type (copper fiber, etc) And of course Cisco makes PoE based phones 7940 7960 which work well with * Grandstream currently requires a wall-wart, but later models are suppose to use PoE as well. I'd personally put the phones on their own subnet so that ACL filtering at the router will be easier, static IP alloc will be easier. hope this helps john brown chagres technologies, inc sip: [EMAIL PROTECTED] ptsn: (01) 505 830 1200 USA On Sun, Aug 17, 2003 at 12:44:43PM -0500, Mike Ciholas wrote: Hi all, I'm looking for recommendations on ethernet switches for a new install. Ideally would want switches with at least 24 ports, ideally with a GE uplink, and that support PoE (power over ethernet) on every port. I've seen lots of switches, and lots of power hubs, but the combination, which makes a lot of sense, seems rare. What is out there? Do the switches need to be special for IP phones in anyway? QoS support? Managed? Also, are there PoE phones that work with *? Most I look at seem to be powered from AC wall blocks. We'd like to centralize the switching and power and provide a UPS so the phone system works when the power goes out. [Apologies, I'm new to this whole concept of IP phones and *.] -- Mike Ciholas(812) 476-2721 voice CIHOLAS Enterprises (812) 476-2881 fax 2626 Kotter Ave, Unit D [EMAIL PROTECTED] Evansville, IN 47715http://www.ciholas.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner at One Unified, and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BudgeTone NAT issues
Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my POTS backhaul. On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig deeply enough to be greatly more specific. However, I do know that: Using NuFone as backhaul, the BudgeTone works fine except in the cases of the called party's line being busy. In this case, I don't get any busy indication, and eventually the call times out at the NuFone end, all the while giving the user at the instrument a ring indication. Using the BudgeTone with iconnecthere, outgoing calls set up just fine, and I get call progress messages from asterisk. The connection dies immediately the called party picks up, with a SIP 486 Temporarily Not Available error. The phone appears to work mostly normally with Vonage, EXCEPT it appears that the RTP stream has to have something fed into it at the calling end; i.e. the called party cannot hear me, nor can I hear the other end, until *I* have said something once the call picks up. Then conversation proceeds normally. I don't have these issues with either the ATA186 nor the TDM20. FYI. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HP300 phone
Hiyas.. Have any of you tried the HP300 phone and got it to work with asterisk ? (sip) /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail cliping digits via sip
At 09:02 PM 8/16/2003 -0600, you wrote: installed about a week ago. I'll go pull to see if there is more recent code I had this problem two weeks ago, and it seems to be gone as of yesterday. However, I did end up changing two things at once: I used a newer CVS version, and I changed the DTMF mode in sip.conf. As far as I know either one might have solved the problem, so if CVS doesn't help, try changing the DTMF mode as well. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk
Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus which I'd connect to asterisk using chan_capi. Now to place a call through asterisk I'd first have to dial into asteriks and then have asterisk give me a dialtone and recognice dtmf. Is there any way to just prefix the phonnumber with lets say a 9 and then have the PBX transmit the rest of the number to asteriks so that I would not have to go through two dialtones? Any suggestions on what PBX to use if this is possible? It's supposed to be for a private home so I should not be some really expensive professional one... Thank you very much! Regards Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail cliping digits via sip
What did you change the DTMF mode to? Where can I find documentation with all the possible options in the config files? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Sunday, August 17, 2003 4:06 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail cliping digits via sip At 09:02 PM 8/16/2003 -0600, you wrote: installed about a week ago. I'll go pull to see if there is more recent code I had this problem two weeks ago, and it seems to be gone as of yesterday. However, I did end up changing two things at once: I used a newer CVS version, and I changed the DTMF mode in sip.conf. As far as I know either one might have solved the problem, so if CVS doesn't help, try changing the DTMF mode as well. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no incoming packets Sound: Recording overrun
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: Hello, and thank you for registering at gnophone.com. Your login information is listed below: Username: miernik Password: *** IAX Phone Number: 17002916107 Please login as soon as possible to http://x.linux-support.net/directory/ to complete the registration phase and activate your gnophone account. Hi, I'd be grateful if any of recipiens of this message, could give me some clues on this problem, as googling the Internet didn't give me any clues. I am a Debian GNU/Linux user and have just installed Gnophone from http://packages.debian.org/unstable/sound/gnophone.html I have gnophone 0.2.4+cvs.20020624-4 I have registered at http://www.gnophone.com/directory/createAccount.php and received the IAX phone number above. I did login, and enter my data in the gnophone program Preferences/Telephone Settings as follows: Use Asterix Server: iaxtel.com Port: 5036 Context: iaxtel Prefix: Username: miernik Password: (same as above) Peer(optional): miernik Sercret(optional) (same as above) I am unable to call anywhere. If I try to call any 1700xxx number, the program sends packets like this (output of tethereal): 20:02:31.6224 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2262 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2355 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2902 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.5429 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:34.4928 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 but no packets ever come back. szrenica.ctnet.pl (212.126.24.133) is my computer. These are the only packets I see. No packets are blocked on any firewall between me and the Internet, my computer is reachable from the internet, you can ping it from the internet now to check. UDP is not blocked. I can even log over SSH to the only firewall between me and the internet (I'm the admin there) and I can see on the external interface that there are also no incoming UDP packets to/from port 5036. Also in my logs I get messages like this: Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Any help would be apprecieated. I'm leaving gnophone running, in case anyone would try to call me. BTW: Duging subscription th gnophone/iaxtel service, I checked the box to be subcribed to gnophone users mailing list, but I didn't get any info how to post to that list. regards, Jan Macek -- Miernik jabberid:[EMAIL PROTECTED] __ ICQ: 4004001 ___/__ tel: +48608233394 __/ mailto:[EMAIL PROTECTED] Sing a declaration against US invasion in Iraq: http://www.moveon.org/declaration/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk
Why not get IP phones and use Asterisk as the PBX from the start?? Will probably save you a LOT of money and many headaches... Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus which I'd connect to asterisk using chan_capi. Now to place a call through asterisk I'd first have to dial into asteriks and then have asterisk give me a dialtone and recognice dtmf. Is there any way to just prefix the phonnumber with lets say a 9 and then have the PBX transmit the rest of the number to asteriks so that I would not have to go through two dialtones? Any suggestions on what PBX to use if this is possible? It's supposed to be for a private home so I should not be some really expensive professional one... Thank you very much! Regards Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail cliping digits via sip
At 04:14 PM 8/17/2003 -0400, you wrote: What did you change the DTMF mode to? Where can I find documentation with all the possible options in the config files? Check out the demo config files - I think those are the only complete documentation in existence. It looks like I actually removed the line dtmfmode=inband from the config in sip.conf. I don't know what the default is. From the demo sip.conf: ;[snomsip] #type=friend #secret=goldfish #host=dynamic #dtmfmode=inband; Choices are inband, rfc2833, or info #defaultip=208.19.48.149 #mailbox=1234,2345 ; Mailbox for message waiting indicator Here is a successful config for using the X-Lite softphone. I am using the 8/16/2003 CVS. [ernest] type=friend username=ernest secret=xxx host=dynamic qualify=1000; Consider it down if it's 1 second to reply defaultip=xxx.xxx.xxx.xxy mailbox=1234 [ryanw] type=friend username=ryanw secret=xxx host=dynamic qualify=1000; Consider it down if it's 1 second to reply defaultip=xxx.xxx.xxx.xxz mailbox=1235 --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail cliping digits via sip
I have removed all the dmtfmode= statements from my sip.conf to begin with. Earlier today I downloaded and compiled the latest CVS and from my testing right now it seems to work a lot better. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger Sent: Sunday, August 17, 2003 4:24 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Voicemail cliping digits via sip At 04:14 PM 8/17/2003 -0400, you wrote: What did you change the DTMF mode to? Where can I find documentation with all the possible options in the config files? Check out the demo config files - I think those are the only complete documentation in existence. It looks like I actually removed the line dtmfmode=inband from the config in sip.conf. I don't know what the default is. From the demo sip.conf: ;[snomsip] #type=friend #secret=goldfish #host=dynamic #dtmfmode=inband; Choices are inband, rfc2833, or info #defaultip=208.19.48.149 #mailbox=1234,2345 ; Mailbox for message waiting indicator Here is a successful config for using the X-Lite softphone. I am using the 8/16/2003 CVS. [ernest] type=friend username=ernest secret=xxx host=dynamic qualify=1000; Consider it down if it's 1 second to reply defaultip=xxx.xxx.xxx.xxy mailbox=1234 [ryanw] type=friend username=ryanw secret=xxx host=dynamic qualify=1000; Consider it down if it's 1 second to reply defaultip=xxx.xxx.xxx.xxz mailbox=1235 --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MORE Questions regarding CDR's
How do I get the asterisk compile to produce the cdr_mysql.so module, assuming this is what I need to get CDR's into my mysql database. Is load = cdr_mysql.so what I put in the modules.conf file as it says? It looks like asterisk loads the modules in /usr/lib/asterisk/modules, but there is no cdr_sql.so module there. Also, why is there a Makefile in /usr/src/asterisk/cdr? Thanks, and sorry for my confusion! Thanks, Scott -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of diana Sent: Saturday, August 16, 2003 6:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Questions regarding CDR's Hi- (b) If I enable the CDR SQL module (to use mysql), does it disable the text logging at the same time? No, as long as you still load the cdr_csv.so module. Thanks Scott S Diana the fat one : ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MORE Questions regarding CDR's
On Sunday 17 August 2003 17:25, Scott Stingel wrote: How do I get the asterisk compile to produce the cdr_mysql.so module, assuming this is what I need to get CDR's into my mysql database. You need to install the mysql client libraries and headers. Follow the appropriate instructions for your distribution to accomplish this. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no incoming packets Sound: Recordingoverrun
An 'ISDN' phone? You mean a handset that actually support ISDN? Didn't know they had these, and if they do I'm sure they wouldn't be cheap. Are you talking BRI or PRI? I'm guessing BRI which means you're right, there is no 'card' to go in an Asterisk box that will do this. However, you might want to look at getting a SIP/MGCP/H323 Gateway that supports BRI-ISDN. There are some out there and personally I'd got for SIP. That way u can terminate/originate calls straight from the Asterisk box. Save a few headache with interop to the other inferrior PABX. :) J On Sun, 17 Aug 2003 22:12:30 +0200 Miernik [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: Hello, and thank you for registering at gnophone.com. Your login information is listed below: Username: miernik Password: *** IAX Phone Number: 17002916107 Please login as soon as possible to http://x.linux-support.net/directory/ to complete the registration phase and activate your gnophone account. Hi, I'd be grateful if any of recipiens of this message, could give me some clues on this problem, as googling the Internet didn't give me any clues. I am a Debian GNU/Linux user and have just installed Gnophone from http://packages.debian.org/unstable/sound/gnophone.html I have gnophone 0.2.4+cvs.20020624-4 I have registered at http://www.gnophone.com/directory/createAccount.php and received the IAX phone number above. I did login, and enter my data in the gnophone program Preferences/Telephone Settings as follows: Use Asterix Server: iaxtel.com Port: 5036 Context: iaxtel Prefix: Username: miernik Password: (same as above) Peer(optional): miernik Sercret(optional) (same as above) I am unable to call anywhere. If I try to call any 1700xxx number, the program sends packets like this (output of tethereal): 20:02:31.6224 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2262 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2355 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2902 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.5429 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:34.4928 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 but no packets ever come back. szrenica.ctnet.pl (212.126.24.133) is my computer. These are the only packets I see. No packets are blocked on any firewall between me and the Internet, my computer is reachable from the internet, you can ping it from the internet now to check. UDP is not blocked. I can even log over SSH to the only firewall between me and the internet (I'm the admin there) and I can see on the external interface that there are also no incoming UDP packets to/from port 5036. Also in my logs I get messages like this: Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Any help would be apprecieated. I'm leaving gnophone running, in case anyone would try to call me. BTW: Duging subscription th gnophone/iaxtel service, I checked the box to be subcribed to gnophone users mailing list, but I didn't get any info how to post to that list. regards, Jan Macek -- Miernik jabberid:[EMAIL PROTECTED] __ ICQ: 4004001 ___/__ tel: +48608233394 __/ mailto:[EMAIL PROTECTED] Sing a declaration against US invasion in Iraq: http://www.moveon.org/declaration/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] no incoming packets Sound: Recordingoverrun
I'm not near my * box at the moment, so can't check this, but IAXTEL isn't down again, is it? Can you ping iaxtel.com. J On Sun, 17 Aug 2003 22:12:30 +0200 Miernik [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: Hello, and thank you for registering at gnophone.com. Your login information is listed below: Username: miernik Password: *** IAX Phone Number: 17002916107 Please login as soon as possible to http://x.linux-support.net/directory/ to complete the registration phase and activate your gnophone account. Hi, I'd be grateful if any of recipiens of this message, could give me some clues on this problem, as googling the Internet didn't give me any clues. I am a Debian GNU/Linux user and have just installed Gnophone from http://packages.debian.org/unstable/sound/gnophone.html I have gnophone 0.2.4+cvs.20020624-4 I have registered at http://www.gnophone.com/directory/createAccount.php and received the IAX phone number above. I did login, and enter my data in the gnophone program Preferences/Telephone Settings as follows: Use Asterix Server: iaxtel.com Port: 5036 Context: iaxtel Prefix: Username: miernik Password: (same as above) Peer(optional): miernik Sercret(optional) (same as above) I am unable to call anywhere. If I try to call any 1700xxx number, the program sends packets like this (output of tethereal): 20:02:31.6224 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2262 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2355 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2902 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.5429 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:34.4928 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 but no packets ever come back. szrenica.ctnet.pl (212.126.24.133) is my computer. These are the only packets I see. No packets are blocked on any firewall between me and the Internet, my computer is reachable from the internet, you can ping it from the internet now to check. UDP is not blocked. I can even log over SSH to the only firewall between me and the internet (I'm the admin there) and I can see on the external interface that there are also no incoming UDP packets to/from port 5036. Also in my logs I get messages like this: Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Any help would be apprecieated. I'm leaving gnophone running, in case anyone would try to call me. BTW: Duging subscription th gnophone/iaxtel service, I checked the box to be subcribed to gnophone users mailing list, but I didn't get any info how to post to that list. regards, Jan Macek -- Miernik jabberid:[EMAIL PROTECTED] __ ICQ: 4004001 ___/__ tel: +48608233394 __/ mailto:[EMAIL PROTECTED] Sing a declaration against US invasion in Iraq: http://www.moveon.org/declaration/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recomendations for an ISDN-PBX to usewith asterisk
Bugga, it's definately a monday. Replied to the wrong subject. (see below). J On Mon, 18 Aug 2003 09:42:05 +1000 Jamie Carl [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* An 'ISDN' phone? You mean a handset that actually support ISDN? Didn't know they had these, and if they do I'm sure they wouldn't be cheap. Are you talking BRI or PRI? I'm guessing BRI which means you're right, there is no 'card' to go in an Asterisk box that will do this. However, you might want to look at getting a SIP/MGCP/H323 Gateway that supports BRI-ISDN. There are some out there and personally I'd got for SIP. That way u can terminate/originate calls straight from the Asterisk box. Save a few headache with interop to the other inferrior PABX. :) J On Sun, 17 Aug 2003 22:12:30 +0200 Miernik [EMAIL PROTECTED] wrote: *This message was transferred with a trial version of CommuniGate(tm) Pro* On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote: Hello, and thank you for registering at gnophone.com. Your login information is listed below: Username: miernik Password: *** IAX Phone Number: 17002916107 Please login as soon as possible to http://x.linux-support.net/directory/ to complete the registration phase and activate your gnophone account. Hi, I'd be grateful if any of recipiens of this message, could give me some clues on this problem, as googling the Internet didn't give me any clues. I am a Debian GNU/Linux user and have just installed Gnophone from http://packages.debian.org/unstable/sound/gnophone.html I have gnophone 0.2.4+cvs.20020624-4 I have registered at http://www.gnophone.com/directory/createAccount.php and received the IAX phone number above. I did login, and enter my data in the gnophone program Preferences/Telephone Settings as follows: Use Asterix Server: iaxtel.com Port: 5036 Context: iaxtel Prefix: Username: miernik Password: (same as above) Peer(optional): miernik Sercret(optional) (same as above) I am unable to call anywhere. If I try to call any 1700xxx number, the program sends packets like this (output of tethereal): 20:02:31.6224 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2262 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2355 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.2902 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:33.5429 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 20:02:34.4928 szrenica.ctnet.pl - 12.37.165.130 UDP Source port: 5036 Destination port: 5036 but no packets ever come back. szrenica.ctnet.pl (212.126.24.133) is my computer. These are the only packets I see. No packets are blocked on any firewall between me and the Internet, my computer is reachable from the internet, you can ping it from the internet now to check. UDP is not blocked. I can even log over SSH to the only firewall between me and the internet (I'm the admin there) and I can see on the external interface that there are also no incoming UDP packets to/from port 5036. Also in my logs I get messages like this: Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Aug 17 21:04:31 szrenica kernel: Sound: Recording overrun Any help would be apprecieated. I'm leaving gnophone running, in case anyone would try to call me. BTW: Duging subscription th gnophone/iaxtel service, I checked the box to be subcribed to gnophone users mailing list, but I didn't get any info how to post to that list. regards, Jan Macek -- Miernik jabberid:[EMAIL PROTECTED] __ ICQ: 4004001 ___/__ tel: +48608233394 __/ mailto:[EMAIL PROTECTED] Sing a declaration against US invasion in Iraq: http://www.moveon.org/declaration/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Regards, Jamie Carl Jazz Inc. Email: [EMAIL PROTECTED] Web:www.jazz-inc.net Phone: +61-414-365-466 Jabber: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cordless SIP phones
Does anyone have any recommendations for a cordless phone that uses SIP (or IAX)? It doesn't have to use 802.11b, but that would be appreciated. Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Recomendations for an ISDN-PBX to use with asterisk
Quoting Oliver Brandt: Hi, I'm planning to buy a new ISDN-PBX (I hope this is the right term for an ISDN phone system). I would also like to connect it to asterisk. As far as I know there is no ISDN card where I can connect an ISDN-Phone to directly working together with asterisk (please correct me if I'm wrong). So what I was thinking of doing was to get a regular ISDN PBX and add a second internal S0 bus which I'd connect to asterisk using chan_capi. Now to place a call through asterisk I'd first have to dial into asteriks and then have asterisk give me a dialtone and recognice dtmf. Is there any way to just prefix the phonnumber with lets say a 9 and then have the PBX transmit the rest of the number to asteriks so that I would not have to go through two dialtones? Any suggestions on what PBX to use if this is possible? It's supposed to be for a private home so I should not be some really expensive professional one... I though this sounded like a good idea too, so I looked around for a small PBX in the UK. First problem was that they are like rocking horse manure here, and the few that you can get are quite expensive. Undeterred I decided to look abroad, and picked one up for about EUR120 or so while on a visit to Munich. It is called a TELNET WILLI (made by Telebau I think) and has 6 analogue extensions plus an internal S0 bus (required for connection to *) - seemed perfect (also kapejod mentioned that he had used one of these). Unfortunately, the PBX is designed purely for the German market and so there is NO English documentation at all. As my German is VERY basic, I've had a really tough time getting it configured (hours of cutting and pasting into babelfish - German technical terms do not translate well :( ). After extensive hair tearing I did manage to get the PBX to work and I've even managed to make and recieve calls through * with it, but there are still niggles that stop me using it properly. I think the impedence might be wrong for UK handsets (something about the sound isn't right). The double dialtone thing is annoying (it does have a Baby Call function that could be used to get around this, but I've run out of enthusiasm for it at the moment). The dialtone and ring are unfamiliar and are not user configurable. The power supply is brick style with a 2 pin euro plug which makes adapting it for UK use difficult. After all the effort, I've decided to shelve it at for the moment and try some 7940's from Ebay (around ?90 + ?20 for a PSU) or the new FXS card from Digium instead. Jamie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cordless SIP phones
On Sun, 2003-08-17 at 17:55, Nathan wrote: Does anyone have any recommendations for a cordless phone that uses SIP (or IAX)? It doesn't have to use 802.11b, but that would be appreciated. I think you're only solution is going to be the Cisco ATA-186, an analog-to-SIP device. Or, you could use the SIP software from TheKompany for the Sharp Zaurus PDA. :) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cordless SIP phones
Well, there is the Netlink phone line from Spectralink: http://www.spectralink.com/products/nl-wts.html My wireless bridge supported prioritizing traffic from spectralink phones, and they apparently will do SIP (and are 802.11b phones), but those are the only cordless phones I know of that will do SIP :( I was hoping for a recommendation/suggestion from someone. Nathan On 17 Aug 2003, Steve Meyers wrote: On Sun, 2003-08-17 at 17:55, Nathan wrote: Does anyone have any recommendations for a cordless phone that uses SIP (or IAX)? It doesn't have to use 802.11b, but that would be appreciated. I think you're only solution is going to be the Cisco ATA-186, an analog-to-SIP device. Or, you could use the SIP software from TheKompany for the Sharp Zaurus PDA. :) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BudgeTone NAT issues
Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my POTS backhaul. On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig deeply enough to be greatly more specific. However, I do know that: Using NuFone as backhaul, the BudgeTone works fine except in the cases of the called party's line being busy. In this case, I don't get any busy indication, and eventually the call times out at the NuFone end, all the while giving the user at the instrument a ring indication. I am uncertain that this particular problem you describe is a NuFone-specific problem. I have seen similar problems with my own IAX2 connections which don't involve NuFone, and I've had circumstances where calls fail at the PSTN side but IAX2 and/or SIP don't get the message, and continues to ring in my ear despite the Zap line on the other end having hung up. I am currently swamped with bug tasks, so I suspect it will be some time before I narrow this problem down and submit an official report with full diagnosis. Anyone else is welcome to the task. :) JT Using the BudgeTone with iconnecthere, outgoing calls set up just fine, and I get call progress messages from asterisk. The connection dies immediately the called party picks up, with a SIP 486 Temporarily Not Available error. The phone appears to work mostly normally with Vonage, EXCEPT it appears that the RTP stream has to have something fed into it at the calling end; i.e. the called party cannot hear me, nor can I hear the other end, until *I* have said something once the call picks up. Then conversation proceeds normally. I don't have these issues with either the ATA186 nor the TDM20. FYI. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No voice call from H.323-phone to SIP-phone
Hi, Did anyone have any comments on the below problem - or did you (shong ching) manage to solve this? I have the same issue - any assistance would be great. Thanks. Regards, Steven Thomas shong ching [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: [Asterisk-Users] No voice call from H.323-phone to SIP-phone .digium.com 12-08-03 05:43 PM Please respond to asterisk-users Hi lists, I am trying to connect SIP Phone and H323 Phone. I can call to from SIP-Phone to H323 with clear voice. But I can't hear the voice calling from H323-phone to SIP-phone. The ring and hookup function is OK. I am using chan_h323 driver. I also tried changing codecs, g711u and g723.1. The result is same. My phones are no branded Taiwanese. I installed pwlib 1.5.0, openh323 1.12.0. H.323-phone is fastconnect mode. NetMeeting works both call. It's not using fastconnect mode. Could I have some suggestions? Regards, Shong Ching ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Festival 1.4.3
I too tried to get festival 1.4.3 working, but no luck. It appeared to compile just fine. The server started okay. I don't have a sound card on the server so I can't test it without *. When I invoke the * festival application, the asterisk process gets 100% CPU time with no sound. I need to stop the process -- it eventually responds to a stop now. I could not get festival 1.4.2 to compile on RH9.0. The festival-1.4.2.diff file does work with festival 1.4.3. Any suggestions would be great. Until then, users will have to listen to my voice. Version information: OS: Redhat 9.0 Asterisk CVS: 8/8/03 21:17 Festival: 1.4.3 Speech Tools: 1.2.3 Asterisk Experience: 1 month ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call waiting
My setup dial 100 - phone - Asterisk - iax - asterisk - x100p if I dial 100 I get a dial tone on the far x100 I need to be able to flash the x100p card over the internet, when I press flash, then dial *0 and flash the x100p On Sat, 2 Aug 2003, Martin Pycko wrote: Well when you use your phone line and you hear the call waiting sound you can press flash on your phone and then *0 and that will generate the flash on your phone line. This switch to the incoming call. regards Martin On Sat, 2 Aug 2003, lists wrote: I have a x100p card that has call waiting on the line comming into it and then into *. is there any way i can use call waiting on that line? Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Chan_h323 one way audio .digium.com 18-08-03 12:19 PM Please respond to asterisk-users i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_h323 one way audio
set ipTos=lowdelay in oh323.conf and try to see what happens. (of course this would mean your switch should have the ability to detect TOS bits in the packet headers) what version of * are you using? did you check against cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, August 18, 2003 11:30 AM Subject: Re: [Asterisk-Users] Chan_h323 one way audio not sure what you mean by 'are you running cvs'? What does the TOS setting do? Regards, Steven Thomas Kelvin Chua [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent by: cc: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] Chan_h323 one way audio .digium.com 18-08-03 12:19 PM Please respond to asterisk-users i also encountered this problem i'm not too sure either but i don't think codec has to do anything with it for i tried mix and matching but to no avail. so for the meantime, try adjusting the tos for oh323 and i think you could live with it by the way, are you running cvs? - Original Message - From: Steven Thomas [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, August 17, 2003 8:56 PM Subject: [Asterisk-Users] Chan_h323 one way audio Hi, I have been using chan_oh323 with a latency issue even on the same network. I am now trying chan_h323 and can only get one way audio. I am testing using SJPhone - SJPhone, and also SJPhone - 7960 (SIP). Any ideas? Must be something obvious that I am missing? Thanks. Regards, Steven Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
