Re: [asterisk-users] Asterisk 1.4: encryption support
Brian Capouch wrote: What's the status of SRTP? I remember seeing things floating around about it being under development, but various sotto voce conversations I've had around over the past few days would indicate that it hasn't gained much/any traction. I'd be glad to be disabused of that notion. I don't think that code has been touched in quite some time. There has been a patch for it for a good while, but it has never merged since it wasn't much use without TLS support for SIP. There has been some good progress towards TCP and TLS support for SIP in the past six months, and I'm hoping to get it finished up by the end of the year. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ --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] Large voicemail
El Thursday 25 October 2007 11:31:37 Tilghman Lesher escribió: On Thursday 25 October 2007 07:40:06 Pepo wrote: I am trying to use Asterisk as the voicemail system of the TELCO where I work. I wanna test with 2 mail boxes ( and later with a better machine/server I hope try with 7 ). How do I include in voicemail.conf the file with the mail boxes?, In a big system like this,is better use text files or any database? Well, if it's the same format, you can use #include. However, with a large system like that, I would tend to use the database to configure mailboxes, while using sound files directly on disk. Until now always use plain text files, How do I can use a Database in the voicemail system? -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --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] Large voicemail
El Thursday 25 October 2007 11:31:37 Tilghman Lesher escribió: On Thursday 25 October 2007 07:40:06 Pepo wrote: I am trying to use Asterisk as the voicemail system of the TELCO where I work. I wanna test with 2 mail boxes ( and later with a better machine/server I hope try with 7 ). How do I include in voicemail.conf the file with the mail boxes?, In a big system like this,is better use text files or any database? Well, if it's the same format, you can use #include. However, with a large system like that, I would tend to use the database to configure mailboxes, while using sound files directly on disk. Until now always use plain text files, How do I can use a Database in the voicemail system? http://www.voip-info.org/wiki-Asterisk+RealTime http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail ___ --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] Treating T1 as trunk in/out, not individual lines
On Saturday 27 October 2007 14:35:55 Benny Amorsen wrote:
DL == Doug Lytle [EMAIL PROTECTED] writes:
DL Michelle Dupuis wrote:
Ok - that's great. I see how the destination number will match to
the exten value, but how do I access the from number '248xxx'?
DL exten = s,1,GotoIf($[${CALLERID(number)} = 248xxx ]?2:3)
That works, of course, but there's also the traditional
ex-gf-function:
exten = s/248XXX,1,NoOp(Matched 248...)
exten = s/X!,1,NoOp(Didn't match)
You can just as easily do the default without any match at all, i.e.
exten = s/248XXX,1,NoOp(Matched 248...)
exten = s,1,NoOp(Didn't match)
Looks cleaner, too.
--
Tilghman
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Re: [asterisk-users] Uniden UIP200 phones
Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Large voicemail
On Sun, 2007-10-28 at 02:27 -0500, Pepo wrote: El Thursday 25 October 2007 11:31:37 Tilghman Lesher escribió: On Thursday 25 October 2007 07:40:06 Pepo wrote: I am trying to use Asterisk as the voicemail system of the TELCO where I work. I wanna test with 2 mail boxes ( and later with a better machine/server I hope try with 7 ). How do I include in voicemail.conf the file with the mail boxes?, In a big system like this,is better use text files or any database? Well, if it's the same format, you can use #include. However, with a large system like that, I would tend to use the database to configure mailboxes, while using sound files directly on disk. Until now always use plain text files, How do I can use a Database in the voicemail system? With Asterisk 1.4 you have a choice to compile voicemail with odbc support. Not sure if that's for storing the voicemail only or for the config too. Regards, Patrick ___ --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] Uniden UIP200 phones
Philipp Kempgen wrote: Lyle Giese wrote: I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=http://127.0.0.1/Bellcore-dr2) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=http://127.0.0.1/Bellcore-dr2) SIPAddHeader(Alert-Info: ...); Regards, Philipp Kempgen Took me a while to notice the difference between - and _ But it works now! Thanks, Lyle ___ --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] Getting SIP Response Code from HANGUPCAUSE
Ah jeez. All I wanted to do was connect to a carrier and then perform fail over logic based on their SIP response. Not supposed to be difficult. This is what Asterisk is supposed to be good at. We have a SIP module, why not have SIP responses available to the module. Now, I have to look at the lossy HANGUPCAUSE variable and make a best guess. Not an ideal situation. We're trying to improve the ASR's we get from providers. They are low, and often they fail calls for no particular reason. They all do it, even the big ones like Verizon. Checking their responses for purpose of trying another carrier on the fly, and reporting is pretty critical. Doug. - Original Message From: Raj Jain [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 27, 2007 11:29:21 AM Subject: Re: [asterisk-users] Getting SIP Response Code from HANGUPCAUSE The only place where it is reasonable to customize is in the specification of the channel in the configuration file. That is where you would customize, for example, whether DTMF is inband, SIP INFO, or RFC 2833, as well as what codecs will be negotiated for that particular user/peer. But you already have the SIP_HEADER function, which is quite contradictory to what you say. This allows users who know what they are doing to examine headers directly. We use this a lot. What would be the harm in having a SIP_RESPONSE function or something alike? I'd agree that SIP response code should be accessible from the dial plan. Knowing the exact SIP response code could be critical for making call processing decisions. The conversion of SIP response codes to Q.931 codes (HANGUPCAUSE) is just too lossy. Building a truly protocol agnostic dial plan API is a worthy goal. But, I think it is somewhat of an unsolvable problem. The signaling protocols are very different and for various reasons people have always wanted access to native information elements carried in the protocol. Perhaps, a very simple solution for this problem could be to support a keyword such as TOPLINE in the SIP_HEADER function to fetch the topmost line in a SIP message. This will not only get the caller the response code for SIP response messages, but will also have the nice byproduct of making the Request-URI available if the message in question is a SIP request. - Raj ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] E4 Superframe EM?
Richard Lyman wrote: Steve Totaro wrote: Richard Lyman wrote: Steve Totaro wrote: I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe EM. I have done EM wink but have no idea about E4 Superframe EM and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve use this zaptel.conf span=1,1,0,d4,ami em=1-24 ; 1-32 for E4 zapata.conf signalling=em channel = 1-24 ; 1-32 for E4 Thanks to everyone who has responded so quickly to my question. To my way of thinking, it would be better to have the legacy tie-line reconfigured to use esf if possible. Is D4 (superframe) well supported in Asterisk, are there less features? If it is virtually the same, then I guess I will just setup Asterisk to use it rather than messing with the legacy system. Thanks, Steve i need sleep that is for sure. the 1-32 for E4 stuff was meant to be for E1 stuff you can just increase the channel count. (1-31) if possible, yes, because you get the added error information with ESF. using d4,ami with em signalling has worked for years for us (on 1.2). i have not tested it with 1.4 if i were you, i would 'test' the above settings first, then change to ESF. that way at least you will have something to fall back too. Ok finally setting this up. How do I debug since it is not PRI? I have a Sangoma card and four tie lines. When i try straight through cables, I see the channels starting and stopping but no green light on card or solid light on proprietary system. Thanks, Steve ___ --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] OT: Managing wireless SIP phone congestion on AP
We are planning a very large Asterisk deployment, using Wifi SIP phones. We've done installs using Spectralink and the SVP to manage congestion at the access points, but we have a client that doesn't want Spectralinks. Anyone have experience with an alternative congestion management (AP association management ?) technology with Asterisk? Anything open source that I'm not aware of? Any other protocols I'm not aware of? Thanks, MD ___ --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] E4 Superframe EM?
Steve Totaro wrote: Richard Lyman wrote: Steve Totaro wrote: Richard Lyman wrote: Steve Totaro wrote: I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe EM. I have done EM wink but have no idea about E4 Superframe EM and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve use this zaptel.conf span=1,1,0,d4,ami em=1-24 ; 1-32 for E4 zapata.conf signalling=em channel = 1-24 ; 1-32 for E4 Thanks to everyone who has responded so quickly to my question. To my way of thinking, it would be better to have the legacy tie-line reconfigured to use esf if possible. Is D4 (superframe) well supported in Asterisk, are there less features? If it is virtually the same, then I guess I will just setup Asterisk to use it rather than messing with the legacy system. Thanks, Steve i need sleep that is for sure. the 1-32 for E4 stuff was meant to be for E1 stuff you can just increase the channel count. (1-31) if possible, yes, because you get the added error information with ESF. using d4,ami with em signalling has worked for years for us (on 1.2). i have not tested it with 1.4 if i were you, i would 'test' the above settings first, then change to ESF. that way at least you will have something to fall back too. Ok finally setting this up. How do I debug since it is not PRI? I have a Sangoma card and four tie lines. When i try straight through cables, I see the channels starting and stopping but no green light on card or solid light on proprietary system. Thanks, Steve How to debug? I have tried straight thru T1, crossover 1-4,2-5, crossover 1-5,2-4, and em_w, em, master, slave, and every combination. Strange thing is that when the CLI was reporting simple switch open/closed is when the other end of the cable was just dangling in the air! I finally thought I was getting somewhere too. Thanks, Steve ___ --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] Read back of caller ID
On Sun, 28 Oct 2007, arkda wrote: I've been looking around for an example of a method of reading back a caller ID value, but I haven't found anything that doesn't use Festival. I'd rather not resort to the Mr. Roboto voice if I can avoid it. Playback of the numbers one at a time is perfectly fine, so I'd like to use the default female Asterisk voice (the sound files are in place on my server). Does anyone have an example of how to accomplish this? SayDigits: -= Info about application 'SayDigits' =- [Synopsis] Say Digits [Description] SayDigits(digits): This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. See the LANGUAGE function for more information on setting the language for the channel. Gordon ___ --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] Read back of caller ID
Knew there must have been an easier way and something I was missing. Thanks
Anthony.
On 10/28/07, Anthony Messina [EMAIL PROTECTED] wrote:
On Saturday 27 October 2007 11:19:05 pm arkda wrote:
I've been looking around for an example of a method of reading back a
caller ID value, but I haven't found anything that doesn't use Festival.
I'd rather not resort to the Mr. Roboto voice if I can avoid it.
Playback of the numbers one at a time is perfectly fine, so I'd like to
use
the default female Asterisk voice (the sound files are in place on my
server). Does anyone have an example of how to accomplish this?
Thanks in advance!
SayDigits(${CALLERID(num)})
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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[asterisk-users] Meet Me sound file
Hi all, I was trying to change some of the sound files for the meet me conference application, the one where the user is waiting in the conference with the users waiting in to join (the M option-- enable music on hold when the conference has a single caller) Also what is the name of this sound file? How do I go about changing the file with some other sound file ? Thanks a lot Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University ___ --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] E4 Superframe EM?
Steve Totaro wrote: Richard Lyman wrote: Steve Totaro wrote: Richard Lyman wrote: Steve Totaro wrote: I need to create a couple of tie lines between a legacy system and an Asterisk system. I was told that the tie lines are E4 Superframe EM. I have done EM wink but have no idea about E4 Superframe EM and Google is not helping me here. Does anyone know about this type of signaling and if Asterisk can handle it? Thanks, Steve use this zaptel.conf span=1,1,0,d4,ami em=1-24 ; 1-32 for E4 zapata.conf signalling=em channel = 1-24 ; 1-32 for E4 Thanks to everyone who has responded so quickly to my question. To my way of thinking, it would be better to have the legacy tie-line reconfigured to use esf if possible. Is D4 (superframe) well supported in Asterisk, are there less features? If it is virtually the same, then I guess I will just setup Asterisk to use it rather than messing with the legacy system. Thanks, Steve i need sleep that is for sure. the 1-32 for E4 stuff was meant to be for E1 stuff you can just increase the channel count. (1-31) if possible, yes, because you get the added error information with ESF. using d4,ami with em signalling has worked for years for us (on 1.2). i have not tested it with 1.4 if i were you, i would 'test' the above settings first, then change to ESF. that way at least you will have something to fall back too. Ok finally setting this up. How do I debug since it is not PRI? I have a Sangoma card and four tie lines. When i try straight through cables, I see the channels starting and stopping but no green light on card or solid light on proprietary system. Thanks, Steve Just for the record. I had to reconfigure the Sangoma startup scripts even though my zap files were correct with d4,ami. The real gotcha was in the cabling. One crossover was 1-5, 2-4, the other 1-4, 2-5 and then straight through all required for the same dialer. I have seen people post T1 crossover as 1-4, 2-4 but I thought they didn't know what they were talking about. Now I know that there are actually two different kinds. Thanks, Steve ___ --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] E4 Superframe EM?
*snipped Just for the record. I had to reconfigure the Sangoma startup scripts even though my zap files were correct with d4,ami. The real gotcha was in the cabling. One crossover was 1-5, 2-4, the other 1-4, 2-5 and then straight through all required for the same dialer. I have seen people post T1 crossover as 1-4, 2-4 but I thought they didn't know what they were talking about. Now I know that there are actually two different kinds. Thanks, Steve sorry about that i just got back in town. I have been lucky with only needing one type of crossover cable. ___ --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] Grandstream 2000 Parking
Hi Is there a way to program #700 to one of the speed dial buttons. I put #700 in the config file for the last button but it's like pressing a dead button. Is there some trick to using the #key to transfer in speed dial buttons. Thanks Kelly ___ --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] SER / Asterisk and mediapath
Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can anyone give me any hints? I wasn't sure if using canreinvite since I wasn't sure if that would affect the caller's interaction in the IVR. Thanks ___ --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] SER / Asterisk and mediapath
On 10/29/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I'm trying to have a SER machine send calls to an Asterisk server working as an IVR. I was able to do this part just fine. Also, when the caller makes certain options in the IVR, the call is then transferred to an extension via SER. This part is also just fine. However, I'm trying to get Asterisk out of the media path once the caller has made a selection in the IVR. Can anyone give me any hints? I wasn't sure if using canreinvite since I wasn't sure if that would affect the caller's interaction in the IVR. Hi yes can canreinvite does the job depends on peer compatability ram ___ --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] Chanspy or Extenspy.
I am usesing this stanz for channel monitoring ;---call monitoring- exten = 996,1,Answer() exten = 996,2,Wait(1) exten = 996,3,ChanSpy(SIP/,q) exten = 996,4,Hangup This is working fine in my case I have asterisk 1.4.x Sanspareils Greenlans [EMAIL PROTECTED] wrote: Sir, I have configured chanspy and extenspy to listen call on any extension but in both case i am unable to hear voice only silence is there. Rajeev. ___ --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 PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Meet Me sound file
On 10/29/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hi all, I was trying to change some of the sound files for the meet me conference application, the one where the user is waiting in the conference with the users waiting in to join (the M option-- enable music on hold when the conference has a single caller) Also what is the name of this sound file? How do I go about changing the file with some other sound file ? Hi look at meetme.conf and you can change the file name and path should be wav file i belive ram ___ --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] Realtime Mysql error
On 10/27/07, wassim darwish [EMAIL PROTECTED] wrote: Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command realtime mysql status on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 minutes, 39 seconds. Can any body help with this; Hi what is the version of asterisk and mysql what distro you are using ram ___ --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] Everyone is busy/congested: IP Trunk
On 10/27/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi Pablo; How the IP address will be wrong, and asterisk able to do registeration on the destination? If the IP address wrong, so I will not be able to register on that IP address. Hi i see 2 causes 1. it could be Dialplan issue ( check how the provider accept the call, 1 or just USA number) 2 provider blocked account check network trace to get more info ngrep should be the ideal tool to check the errors in network trace ram ___ --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] Realtime Mysql error
search , firewall, and confiration the software. the configuration the user is bad ??? use asterisk2billing it is good On 10/28/07, ram [EMAIL PROTECTED] wrote: On 10/27/07, wassim darwish [EMAIL PROTECTED] wrote: Hi: Iam using an asterisk server with astcc ,iam facing a problem with astcc that when the call is hangup sometimes astcc doesnt calculate the call cost and the call time and without writing the call status on cdrs table . I tried to run this command realtime mysql status on the asterisk console and that what i've got: [Oct 27 01:05:32] ERROR[2607]: res_config_mysql.c:637 mysql_reconnect: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. Connected to [EMAIL PROTECTED], port 3306 with username root for 9 hours, 43 minutes, 39 seconds. Can any body help with this; Hi what is the version of asterisk and mysql what distro you are using ram ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registration of Snom 320 phone with Asterisk 1.4.13
Here are more details: The phone and the Asterisk box are behind the same router (the Asterisk machine is 192.168.0.2 and the phone is 192.168.0.4). A ping command works: [EMAIL PROTECTED]:~$ ping -c 10 192.168.0.4 PING 192.168.0.4 (192.168.0.4) 56(84) bytes of data. 64 bytes from 192.168.0.4: icmp_seq=1 ttl=64 time=0.500 ms 64 bytes from 192.168.0.4: icmp_seq=2 ttl=64 time=0.491 ms 64 bytes from 192.168.0.4: icmp_seq=3 ttl=64 time=0.493 ms 64 bytes from 192.168.0.4: icmp_seq=4 ttl=64 time=0.495 ms 64 bytes from 192.168.0.4: icmp_seq=5 ttl=64 time=0.495 ms 64 bytes from 192.168.0.4: icmp_seq=6 ttl=64 time=0.493 ms 64 bytes from 192.168.0.4: icmp_seq=7 ttl=64 time=0.493 ms 64 bytes from 192.168.0.4: icmp_seq=8 ttl=64 time=0.495 ms 64 bytes from 192.168.0.4: icmp_seq=9 ttl=64 time=0.505 ms 64 bytes from 192.168.0.4: icmp_seq=10 ttl=64 time=0.492 ms --- 192.168.0.4 ping statistics --- 10 packets transmitted, 10 received, 0% packet loss, time 9005ms rtt min/avg/max/mdev = 0.491/0.495/0.505/0.014 ms [EMAIL PROTECTED]:~$ However, the phone never appears to receive the responses from Asterisk to its register requests. The error on the phone is: [2]29/10/2007 17:02:59: Transport Error: Pending packet 1046807: generating fake [2]29/10/2007 17:02:59: Registrar [EMAIL PROTECTED] timed out From /etc/asterisk/sip.conf: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to A realm=192.168.0.2 context = default ;Default for incoming calls [5549] disallow=all allow=ulaw allow=alaw allow=gsm type=friend ;(inbound and outbound calls accepted) secret=localphone ; obvious password for testing host=dynamic callerid=Jason White 5549 dtmfmode=auto mailbox=5549 ;(Asterisk VM-system's mailbox #) The output from sip set debug is attached, as captured earlier by the script command. Asterisk version 1.4.13, Debian GNU/Linux Sid (up to date); this phone has successfully registered with external Asterisk servers. Suggestions are much appreciated. --- SIP read from 192.168.0.4:2048 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK-1fk1w5i9tp27;rport From: Jason White sip:[EMAIL PROTECTED];tag=rrzoty966y To: Jason White sip:[EMAIL PROTECTED] Call-ID: 3c33894c8198-jfnao0gkzzkv CSeq: 341 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2048;line=dz9s1zzw;flow-id=1;q=1.0;+sip.instance=urn:uuid:6467e491-3f54-4b76-8777-359712b5e388;audio;mobility=fixed;duplex=full;description=snom320;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO User-Agent: snom320/7.1.8 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.4 Expires: 3600 Content-Length: 0 - --- (14 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.4 : 2048 (NAT) [Kjdc*CLI --- Transmitting (no NAT) to 192.168.0.4:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK-1fk1w5i9tp27;received=192.168.0.4;rport=2048 From: Jason White sip:[EMAIL PROTECTED];tag=rrzoty966y To: Jason White sip:[EMAIL PROTECTED] Call-ID: 3c33894c8198-jfnao0gkzzkv CSeq: 341 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 192.168.0.4:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK-1fk1w5i9tp27;received=192.168.0.4;rport=2048 From: Jason White sip:[EMAIL PROTECTED];tag=rrzoty966y To: Jason White sip:[EMAIL PROTECTED];tag=as06798431 Call-ID: 3c33894c8198-jfnao0gkzzkv CSeq: 341 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=192.168.0.2, nonce=0048861c Content-Length: 0 Scheduling destruction of SIP dialog '3c33894c8198-jfnao0gkzzkv' in 32000 ms (Method: REGISTER) [Kjdc*CLI --- SIP read from 192.168.0.4:2048 --- REGISTER sip:192.168.0.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.4;branch=z9hG4bK-1fk1w5i9tp27;rport From: Jason White sip:[EMAIL PROTECTED];tag=rrzoty966y To: Jason White sip:[EMAIL PROTECTED] Call-ID: 3c33894c8198-jfnao0gkzzkv CSeq: 341 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2048;line=dz9s1zzw;flow-id=1;q=1.0;+sip.instance=urn:uuid:6467e491-3f54-4b76-8777-359712b5e388;audio;mobility=fixed;duplex=full;description=snom320;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO User-Agent: snom320/7.1.8 Supported: gruu Allow-Events: dialog X-Real-IP: 192.168.0.4 Expires: 3600 Content-Length: 0 - --- (14 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.4 : 2048 (NAT) --- Transmitting (no NAT) to 192.168.0.4:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP
[asterisk-users] Users location --help required
Hello all, i am Presently working on integration of asterisk and openser i have a doubt regarding the asterisk . if you take openser when users register it stores the users in location table whether the users running behind NAT or on global ips and when comes to asterisk where does it store ? because i have seen the documentation of integration of asterisk and openser realtime and content there talked about realtime integration of subscriber and sip.conf tables . and i dont want to register users under asterisk so it should fetch the location of users from location table of openser can above fetching mechanism from openser to asterisk using database views be possible? Thanks in advance Srinivas Antarvedi ___ --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] Users location --help required
On 10/29/07, srinivas Antarvedi [EMAIL PROTECTED] wrote: Hello all, i am Presently working on integration of asterisk and openser i have a doubt regarding the asterisk . if you take openser when users register it stores the users in location table whether the users running behind NAT or on global ips and when comes to asterisk where does it store ? because i have seen the documentation of integration of asterisk and openser realtime and content there talked about realtime integration of subscriber and sip.conf tables . and i dont want to register users under asterisk so it should fetch the location of users from location table of openser can above fetching mechanism from openser to asterisk using database views be possible? Hi yes Location table is users registred so you can view them in any database ram ___ --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
