[asterisk-users] Call files with extensions.ael : One app must be specified
Hi, Using a 1.4 system in which dialplan is written using extensions.conf, I can use a custom .call file. On another system in which dialplan is written using extensions.ael, I can't use any custom .call file : system keeps replying : apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/toto.call When I compare both dialplans using CLI dialplan show, I don't see much differences : [ Context 'local' created by 'pbx_ael' ] (in AEL-enabled) [ Context 'local' created by 'pbx_config' ] (in non AEL-enabled) Here is the call file (I also tried commenting out Priority): Channel: SIP/700 CallerID: 692 692 MaxRetries: 1 WaitTime: 60 RetryTime: 5 Context: local Extension: 700 Priority: 1 What shall I edit to have it working ? Regards ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. The problem is, that the old behavior is there since 1.4 and many users use it. Thus, if you change it now you break the dialplan of many users. klaus ___ -- 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] Multiple SIPGate accounts.
I supsect that the incoming request is tried to match against a peer - based on IP:port. Thus, it will always match the same peer, regardsless if the call is incoming from account1 or account2. Try using the same context in both peer definitions and put the 1212121 and 1313131 extensions in this context. regards klaus Razza schrieb: Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8 SNIP! 8 Register = 1212121:a...@sipgate.co.uk/1212121 http://1212121:a...@sipgate.co.uk/1212121 Register = 1313131:b...@sipgate.co.uk/1313131 http://1313131:b...@sipgate.co.uk/1313131 8 SNIP! 8 [sipgate] type=friend username=1212121 secret= host=sipgate.co.uk http://sipgate.co.uk fromuser=1212121 fromdomain=sipgate.co.uk http://sipgate.co.uk nat=yes authuser=1212121 dtmfmode=rfc2833 context=infoline_SG insecure=very canreinvite=no disallow=all allow=alaw [2sipgate2] type=friend username=1313131 secret= host=sipgate.co.uk http://sipgate.co.uk fromuser=1313131 fromdomain=sipgate.co.uk http://sipgate.co.uk nat=yes authuser=1313131 dtmfmode=rfc2833 context=infoline_config_SG insecure=very canreinvite=no disallow=all allow=alaw Not that it really matters as these work when the other account is disabled, Excerpts from extensions.conf - 8 SNIP! 8 [infoline_SG] exten = 1212121,1,Goto(infoline,s,1) [infoline] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(/var/lib/asterisk/infolinesounds/welcomeHL) 8 SNIP! 8 [infoline_config_SG] exten = 1313131,1,Answer exten = 1313131,2,Background(/var/lib/asterisk/infolinesounds/welcomeHLC) exten = 1313131,3,Authenticate(1234) exten = 1313131,4,Goto(infoline_config,s,1) [infoline_config] exten = s,1,Answer exten = s,2,DigitTimeout,5 exten = s,3,ResponseTimeout,10 exten = s,4,BackGround(/var/lib/asterisk/infolinesounds/welcomeHLC) 8 SNIP! 8 ___ -- 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
[asterisk-users] switchtype QSIG and Asterisk implementation
Hi, Is Asterisk fully QSIG-compliant? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by Alcatel. However, I've searched for QSIG + Asterisk on the web and came up with some posts saying that Asterisk may not fully implement QSIG (eg.: http://threebit.net/mail-archive/asterisk-users/msg14000.html). Are the latest Asterisk versions (both 1.2 and 1.4, not to mention 1.6) QSIG-compliant or are there known issues? Is anyone here happily running a LegacyPBX---QSIG_PRI---ASTERISK system? I've read this page and it seems that the author did not succeed in setting up QSIG between Alcatel 4400 and Asterisk: http://www.voip-info.org/wiki/view/Alcatel+4400+via+PRI Thanks, Vieri ___ -- 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] switchtype QSIG and Asterisk implementation
On Wed, Feb 25, 2009 at 01:02:10AM -0800, Vieri wrote: Hi, Is Asterisk fully QSIG-compliant? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. ISDN is implemented is Asterisk and in libpri. What version of libpri do you use? What version of Asterisk, exactly? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] switchtype QSIG and Asterisk implementation
--- On Wed, 2/25/09, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Hi, Is Asterisk fully QSIG-compliant? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. ISDN is implemented is Asterisk and in libpri. What version of libpri do you use? What version of Asterisk, exactly? 1) libpri 1.2.5 with Asterisk 1.2.30 2) libpri 1.4.5 with Asterisk 1.4.21.2 ___ -- 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] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide us VOIP services in the 916 area code. I am wondering how can I get the bandwidth.com service, Thanks, Vikas ___ -- 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] switchtype QSIG and Asterisk implementation
2009/2/25 Vieri rentor...@yahoo.com However, it seems that Alcatel's latest firmware has dropped support for euroisdn 1. That is very surprising as I would classify QSIG to a private PBX-to-private PBX protocol, not a private PBX-to-public ISDN. If this classification is true, it would be a commercial suicide for Alcatel to narrow its targeted customers to those living in countries where Alcatel PBX are mostly sold by Telcos (in think it's the case in Italy but not in France, for instance) as those are the only ones that can change protocol used to interconnect with public ISDN. So if, euroisdn support has been dropped in the PBX you're trying to inconnect with, that may come from the company that installed this PBX and deliberately choosed to drop euroisdn feature, for a reason. 2. Alcatel 4400 is a very old product. I'm also surprised it still gets software updates though it was possible to upgrade it to an OmniPCX which is the current Alcatel PBX. 3. Anyway, how is your current setup ? Public ISDN --BRI or PRI --- Asterisk ---BRI or PRI --- Alcatel 4400 4. From my poor understanding of libpri, Asterisk complies with parts of QSIG standards. I tried hard to find a doc describing Asterisk QSIG features but couldn't find anything. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with Internet connectivity
Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed directly to the PBX without any IP needed, ISDN to ISDN. Once the IP connectivity was reestablished everything worked fine again. So I wonder what could be the reason that Asterisk blocked all the telephony. thanks klaus ___ -- 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] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
On Wed, 25 Feb 2009, Vikas wrote: I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide us VOIP services in the 916 area code. I am wondering how can I get the bandwidth.com service, Why would you persist with a company who can't service your needs when there are others who will? Now not being local to your country, there may be other issues at stake here and I don't know about, but voting with your feet *and* telling bandwidth.com why you've gone to a competitor might work here... (Ye Gods: $30 a month for a SIP trunk! Wish I could charge that!) 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] multiple asterisks in a server
Rilawich Ango schrieb: It seems better to install once with multiple instances. Do we need to take care the port or IP of each instance? of course you have to. On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Klaus Darilion wrote: Rilawich Ango wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Just to have several Asterisk instances on a single server you do not need to install it multiple times. Install it once and start it multiple times. Of course you have to have a dedicated configuration for each server, eg: /etc/asterisk/instance1/* /etc/asterisk/instance2/* /etc/asterisk/instance3/* Then you start the Asterisk process and specify the location of the asterisk.conf file. asterisk -C /etc/asterisk/instance1/asterisk.conf asterisk -C /etc/asterisk/instance2/asterisk.conf asterisk -C /etc/asterisk/instance3/asterisk.conf Further, in asterisk.conf specify for each asterisk instance a different location of: spool directory, PID file, btw: I use a common /var/lib/asterisk/ as I want to have the same sounds for all instances. This gives a problem when you use 1.4, as 1.4 can not configure the location of astdb. For these you have to apply this patch: http://bugs.digium.com/view.php?id=14257 regards klaus ___ -- 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 ___ -- 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] switchtype QSIG and Asterisk implementation
Hi, On Wed, Feb 25, 2009 at 10:02 AM, Vieri rentor...@yahoo.com wrote: Is Asterisk fully QSIG-compliant? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. That is a good question. I had the same dilemma here. Finally I am using my OXE via Q.SIG but does not know call transfer and other functions are implemented or not. I have found same documents as you. Unfortunately no useful help even here nor Alcatel forums. For making clear I am not bitching and many many thanks and respect for every help I have received from here just I did not received any useful answer on this topic yet. The libpri 1.4.9 have some new features but no details. From changelog: 2008-10-17 16:13 + [r636] Matthew Fredrickson cres...@digium.com * pri.c, pri_internal.h, pri_q931.h, q931.c, pri_facility.c, pri_facility.h, libpri.h: Merging in additional Q.SIG features in #13454. Includes Q.SIG physical/logical channel mapping support, extended coding of Q.SIG name operations (calling name), and call rerouting support via added dialplan application. What added dialplan application means in this context? New parameter to dial? Might a new function? Bye, a ___ -- 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] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
I am wondering how can I get the bandwidth.com service, Why would you persist with a company who can't service your needs when there are others who will? +1 This industry is full of companies staffed by morons who don't give a s*. Then these companies go bust... and the idiot owners wonder why. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Internet connectivity
Klaus Darilion a écrit : Hi! Hallo I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed directly to the PBX without any IP needed, ISDN to ISDN. Once the IP connectivity was reestablished everything worked fine again. So I wonder what could be the reason that Asterisk blocked all the telephony. I'm thinking about a DNS problem which make Atserisk reacting very slow. Especially if you're EP are SIP/IAX Phones. -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Internet connectivity
2009/2/25 Klaus Darilion klaus.mailingli...@pernau.at Hi! I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed directly to the PBX without any IP needed, ISDN to ISDN. Once the IP connectivity was reestablished everything worked fine again. So I wonder what could be the reason that Asterisk blocked all the telephony. thanks klaus Asterisk is using dns resolution, when he is unable to reach dns server * freezes, it's a know issue, this can be avoided if you point asterisk to your local dns service (inside private network) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with Internet connectivity
On 25 Feb 2009, at 10:38, Klaus Darilion wrote: I have a setup with Asterisk in front of a PBX connected with ISDN to the PSTN and to the PBX. This Asterisk (a old 1.2 instance) is doing ENUM for outgoing calls and allows incoming calls per SIP. Recently the IP connectivity for this location was down the whole telephony was down too - not even incoming calls did work. This is really strange as incoming calls from PSTN are routed directly to the PBX without any IP needed, ISDN to ISDN. Once the IP connectivity was reestablished everything worked fine again. So I wonder what could be the reason that Asterisk blocked all the telephony. Look in the log? Recreate the incident and put debugging on? ___ -- 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] trunk to trunk
To Robert Broyles,Thank you very much, it is very helpful information. Regards, Leonid 2009/2/18 Robert Broyles rob...@poornam.com Hi, You might want to check out this tutorial: http://hostseries.com/connecting-to-asterisk-servers-via-sip/ It's a good place to start. -- Regards, Robert Broyles Leonja Cerebro wrote: Hi, Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of Asterisk B (registered in Asterisk A as extension) to incoming call across another trunk of Asterisk B to extension of Asterisk C What the dial plan should be? Thanks -- We never did too much talking anyway So don't think twice, it's all right -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- We never did too much talking anyway So don't think twice, it's all right ___ -- 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] trunk to trunk
Glad I could help!! :-D Leonja Cerebro wrote: To Robert Broyles, Thank you very much, it is very helpful information. Regards, Leonid 2009/2/18 Robert Broyles rob...@poornam.com mailto:rob...@poornam.com Hi, You might want to check out this tutorial: http://hostseries.com/connecting-to-asterisk-servers-via-sip/ It's a good place to start. -- Regards, Robert Broyles Leonja Cerebro wrote: Hi, Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk of Asterisk B (registered in Asterisk A as extension) to incoming call across another trunk of Asterisk B to extension of Asterisk C What the dial plan should be? Thanks -- We never did too much talking anyway So don't think twice, it's all right ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- We never did too much talking anyway So don't think twice, it's all right ___ -- 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 DISCLAIMER : This email and any files transmitted with it are property of Poornam Info Vision Pvt. Ltd. This email contains confidential information intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the sender immediately and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments.___ -- 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] switchtype QSIG and Asterisk implementation
--- On Wed, 2/25/09, Olivier oza-4...@myamail.com wrote: So if, euroisdn support has been dropped in the PBX you're trying to inconnect with, that may come from the company that installed this PBX and deliberately choosed to drop euroisdn feature, for a reason. I trust this company and they are saying that Alcatel suggested us to switch to QSIG because EuroISDN is not supported. This company is asking Alcatel to fix this issue and we're waiting for a feedback. 2. Alcatel 4400 is a very old product. I'm also surprised it still gets software updates though it was possible to upgrade it to an OmniPCX which is the current Alcatel PBX. Our system is an omnipcx enterprise oxe (we still know it as a 4400). 3. Anyway, how is your current setup ? Public ISDN --BRI or PRI --- Asterisk ---BRI or PRI --- Alcatel 4400 Public ISDN -- Asterisk -- PRI -- Alcatel 4400 ___ -- 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] GSM codec is a good choice ???
But in my case, I don't need trascoding because every chanel is in GSM and voicemail has gsm sound files. And for the moment, my Asterisk is not connected to the PSTN, so there is no trascoding gsm-to-PCM or to analog. So I think gsm is a good choice for my scenario, do you ??? Thanks a lot !!! On Wed, Feb 25, 2009 at 5:33 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Tue, Feb 24, 2009 at 11:16:51PM -0200, David fire wrote: out there is a free for educational and no commercial G729 lib for asterisk you can use it to test in a non-comercial system. For personal use? Maybe. For educational use: not really. The licensing of the Intel codec code are not that nice. And naturally, if you wan ta good speech codec with a high quality and yet good compression, and no extra bagage of patents, your first choice should be Speex. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alejandro Cabrera Obed aco1...@gmail.com www.alejandrocabrera.com.ar ___ -- 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] strange text message:)
I don't know what is MWI Message. All I know is that i can find these messages in my SMS inbox and has the sender voicem...@mydomain.xxx On 2/24/09, OCG Technical Support supp...@ocg.ca wrote: Are you sure this is not just a standard SIP MWI message? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S. Sent: February 23, 2009 8:01 PM To: Asterisk Users List Subject: Re: [asterisk-users] strange text message:) is any chance to use this feature to send messages on this kind of phones? On Tue, Feb 24, 2009 at 1:39 AM, David fire ddf...@gmail.com wrote: you are getting the info about the voicemail becausethe soft on your phone support it. in sip.conf you can find some parameters to send that info. in other soft phones like x-lite you will have the same info. David 2009/2/23 Catalin S. jonsonpla...@gmail.com Hello guys, I recently observed that my asterisk sends me sms like messages on my phone (Nokia E71), I mean is SMS but is delivered some kind in-band though VoIP. Is strange because this messages contains informations about my voicemail and is sent by voicem...@mydomainxxx.com. I noticed that this messages appears every time when I logged in with my phone on my sip account. I'm interested about how can I send these messages with other information's or whatever I want to my terminals. Also I observed that works with Nokia E71 only. Maybe is because I updated some software on It , Not Firmware. Do you guys observed this too? Thank you for support. Catalin. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 ___ -- 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
[asterisk-users] usegmtime=yes for cdr_custom
Hi! I have set usegmtime=yes in cdr.conf, but unfortunately this is only for cdr-csv, not for cdr-custom. AFAIS there is no such option for cdr_custom.conf. Is there any workaround to get GMT timestamps in cdr-custom too? thanks klaus ___ -- 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] SIP_CODEC variable
Hi,
I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec
being used. I have exclusively Polycom phones for this test, and basically I
want all communications to use g729 (preferred codec), except for pagine 20
phones (which busts my g729 license count). In that case I want to use gsm.
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate Page command call. But I get this in th CLI:
NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring ${SIP_CODEC}
variable because it is not shared by both ends.
All my registered phones are using g729 and gsm in the sip definitions.
What could it be?
Mike
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Re: [asterisk-users] multiple asterisks in a server
yes, you need to make sure bindaddr is set correctly in iax.conf, sip.conf, dundi.conf, manager.conf and any other files that might include bindaddr for BOTH instances of asterisk, you can't allow one to bind to all ip's and the other just to bind to one - it won't work. 2009/2/25 Rilawich Ango maillist...@gmail.com It seems better to install once with multiple instances. Do we need to take care the port or IP of each instance? - Show quoted text - On Wed, Feb 25, 2009 at 5:36 AM, Klaus Darilion klaus.mailingli...@pernau.at wrote: Klaus Darilion wrote: Rilawich Ango wrote: Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Just to have several Asterisk instances on a single server you do not need to install it multiple times. Install it once and start it multiple times. Of course you have to have a dedicated configuration for each server, eg: /etc/asterisk/instance1/* /etc/asterisk/instance2/* /etc/asterisk/instance3/* Then you start the Asterisk process and specify the location of the asterisk.conf file. asterisk -C /etc/asterisk/instance1/asterisk.conf asterisk -C /etc/asterisk/instance2/asterisk.conf asterisk -C /etc/asterisk/instance3/asterisk.conf Further, in asterisk.conf specify for each asterisk instance a different location of: spool directory, PID file, btw: I use a common /var/lib/asterisk/ as I want to have the same sounds for all instances. This gives a problem when you use 1.4, as 1.4 can not configure the location of astdb. For these you have to apply this patch: http://bugs.digium.com/view.php?id=14257 regards klaus ___ -- 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 ___ -- 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] GSM codec is a good choice ???
2009/2/25 Alejandro Cabrera Obed aco1...@gmail.com But in my case, I don't need trascoding because every chanel is in GSM and voicemail has gsm sound files. And for the moment, my Asterisk is not connected to the PSTN, so there is no trascoding gsm-to-PCM or to analog. So I think gsm is a good choice for my scenario, do you ??? Hi Alejandro! Just to answer your question clearly: Yes, GSM would be a working option for your scenario. If you ever need G711 to connect to ISDN the transcoding should be no problem for a P4 class system and for example 30 ISDN-Lines. But what the others want to say is that buying new phones just to avoid paying for G729 licenses may not be a good idea as the licenses are quite cheap (US$10 for every transcoding you USE at the same time, NOT for every phone you have). I hope that answers your question. Christian ___ -- 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] GSM codec is a good choice ???
On Wed, 25 Feb 2009 09:33:42 +0200, Tzafrir Cohen wrote: On Tue, Feb 24, 2009 at 11:16:51PM -0200, David fire wrote: out there is a free for educational and no commercial G729 lib for asterisk you can use it to test in a non-comercial system. For personal use? Maybe. For educational use: not really. The licensing of the Intel codec code are not that nice. And naturally, if you wan ta good speech codec with a high quality and yet good compression, and no extra bagage of patents, your first choice should be Speex. The trouble with Speex is that it has extremely limited support in hardware. I've yet to see a high quality IP phone that supports Speex directly. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] GSM codec is a good choice ???
On Wed, Feb 25, 2009 at 07:25:10AM -0600, Michael Graves wrote: The trouble with Speex is that it has extremely limited support in hardware. I've yet to see a high quality IP phone that supports Speex directly. OTOH, it's well supported in soft phones. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM codec is a good choice ???
On Wed, 25 Feb 2009 15:46:09 +0200, Tzafrir Cohen wrote: On Wed, Feb 25, 2009 at 07:25:10AM -0600, Michael Graves wrote: The trouble with Speex is that it has extremely limited support in hardware. I've yet to see a high quality IP phone that supports Speex directly. OTOH, it's well supported in soft phones. True, but that doesn't get you very far in most real world installations. IME, soft phones are an accessory to an installation, not typically the focus. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves fwd 54245 ___ -- 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] SIP_CODEC variable
On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate Page command call. But I get this in th CLI:
NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
${SIP_CODEC} variable because it is not shared by both ends.
This is a wild guess (and I don't currently have the time to check it
out properly), but if my memory serves me the Polycom phones don't
support the GSM codec. You might try ulaw instead.
--
Jared Smith
Digium, Inc. | Training Manager
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Re: [asterisk-users] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher wrote:
On Tuesday 24 February 2009 13:44:25 Barry L. Kline wrote:
Here's one that may be of interest to any upgraders. If you rely on the
behavior of gosub you may want to make note of this change.
I have an incoming call context:
exten = _,n,GoSub(incoming,${EXTEN},1(${EXTEN}));
that is supposed to gosub into the incoming extension at priority 1.
Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the
requested extension wasn't present in the incoming context.
When I upgraded to 1.6.0.6 this behavior changed and I would simply get
an error on the console that a matching extension was not found, and the
dialplan would simply stop. It was easy enough to add:
[incoming]
exten = _,1,Goto(i,1)
to restore the previous behavior (I'm looking at four-digits from a PRI)
which I should probably have done anyway.
I don't know if this is a bug or WAD but just wanted to mention it.
It was a bug. Gosub/Goto should NEVER go to the i extension, unless that
target is explicitly given. The use of the i extension for invalid
extensions is limited to WaitExten/Background.
Why should it be so limited? It's clearly not now, and it's not been
considered a bug - certainly no bug reports or user confusion. Some of
us have used this behaviour for quite a while. It's very useful.
Why change?
sean
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Re: [asterisk-users] SIP_CODEC variable
On Wed, 25 Feb 2009, Jared Smith wrote:
On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate Page command call. But I get this in th CLI:
NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
${SIP_CODEC} variable because it is not shared by both ends.
This is a wild guess (and I don't currently have the time to check it
out properly), but if my memory serves me the Polycom phones don't
support the GSM codec. You might try ulaw instead.
True, that. They do G.729 though!
j
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[asterisk-users] Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first pleas for help.
I'm having issues where a parked call using the macro below is getting stuck.
Users park the call via a blfxfer key on an Aastra phone. If the call is a
blind transfer, it tries to park the call. If it isn't a blind transfer, it
tries to unpark the call. Only 2 extensions (2759 and 2760) are doing the
parking. The other extensions only pick up calls (by dialing the 3 digit park
code. The phone shows as in use and there is a call that I see via core show
channels. I can't seem to soft hangup the stuck channel either. Only killing
Asterisk forcefully will solve the issue. We're running Asterisk 1.4.18.
Thanks for any help!
[parallelparking]
exten = _7[89]X,1,Noop(Attempting to parallel park...)
exten = _7[89]X,n,Answer
exten = _7[89]X,n,Set(PARKINGEXTEN=${EXTEN})
exten = _7[89]X,n,GotoIf($[${BLINDTRANSFER} != ]?dopark:dounpark)
exten = _7[89]X,n(dopark),Noop(Going to try to park this call)
exten = _7[89]X,n,Set(RECALLEXTEN=${BLINDTRANSFER:4:4})
exten =
_7[89]X,n,ParkAndAnnounce(PARKED|180|Local/parkedannou...@parallelparking|parkreturn,${RECALLEXTEN},1)
exten = _7[89]X,n,Hangup
exten = _7[89]X,n(dounpark),Noop(Going to try to un-park this call)
exten = _7[89]X,n,ParkedCall(${EXTEN})
exten = _7[89]X,n,Hangup
exten = parkedannounce,1,Noop
exten = parkedannounce,n,Answer
exten = parkedannounce,n,Wait(1)
exten = parkedannounce,n,Hangup
[parkreturn]
exten = _,1,Noop(Returning Parked Call)
exten = _,n,SIPAddHeader(Alert-Info: info=${AASTRA_PARKRINGBACK})
exten = _,n,Set(CALLERID(name)=FrPark:${CALLERID(name)})
exten = _,n,Dial(SIP/${EXTEN},60)
exten = _,n,Hangup
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
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Re: [asterisk-users] Stuck Parked Calls?
BTW, hate to reply to myself, but here is what core show channels shows for
the stuck call:
SIP/2754-0849ce682...@parkreturn:1Up (None)
Also, below is the core show channel on the SIP channel:
-- General --
Name: SIP/2754-0849ce68
Type: SIP
UniqueID: 1235508605.71766
Caller ID: 2754
Caller ID Name: (N/A)
DNID Digits: (N/A)
State: Up (6)
Rings: 0
NativeFormats: 0x4 (ulaw)
WriteFormat: 0x4 (ulaw)
ReadFormat: 0x4 (ulaw)
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 132
Frames in: 0
Frames out: 0
Time to Hangup: 0
Elapsed Time: 17h54m39s
Direct Bridge: none
Indirect Bridge: none
-- PBX --
Context: parkreturn
Extension: 2760
Priority: 1
Call Group: 0
Pickup Group: 0
Application: (N/A)
Data: (None)
Blocking in: (Not Blocking)
Variables:
RTPAUDIOQOS=ssrc=724684267;themssrc=2145401849;lp=0;rxjitter=0.36;rxcount=6;txjitter=0.00;txcount=6;rlp=0;rtt=0.00
RECALLEXTEN=2760
PARKINGEXTEN=792
siptransfer_referer=2...@10.10.220.2
SIPTRANSFER=yes
SIPDOMAIN=10.10.220.2
BLINDTRANSFER=SIP/2760-b2e42b60
BRIDGEPEER=SIP/2760-b2e42b60
DIALEDPEERNUMBER=2754
sipcallid=60001c104135b9967ef3d91d6649c...@10.10.220.2
SIPADDHEADER01=Alert-Info: info=Bellcore-dr4
CDR Variables:
level 1: clid=2760
level 1: src=2760
level 1: dst=792
level 1: dcontext=analog-voip
level 1: channel=SIP/2754-0849ce68
level 1: lastapp=ParkAndAnnounce
level 1:
lastdata=PARKED|180|Local/parkedannou...@parallelparking|parkreturn|2760|1
level 1: start=2009-02-24 14:49:10
level 1: answer=2009-02-24 14:49:14
level 1: end=2009-02-24 14:49:14
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1235508550.71744
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
- Original Message -
From: Jonathan C. Bailey jbai...@co.marshall.ia.us
To: asterisk-users@lists.digium.com
Sent: Wednesday, February 25, 2009 8:39:42 AM GMT -06:00 US/Canada Central
Subject: [asterisk-users] Stuck Parked Calls?
I've lurked for a while, but I think this is one of my first pleas for help.
I'm having issues where a parked call using the macro below is getting stuck.
Users park the call via a blfxfer key on an Aastra phone. If the call is a
blind transfer, it tries to park the call. If it isn't a blind transfer, it
tries to unpark the call. Only 2 extensions (2759 and 2760) are doing the
parking. The other extensions only pick up calls (by dialing the 3 digit park
code. The phone shows as in use and there is a call that I see via core show
channels. I can't seem to soft hangup the stuck channel either. Only killing
Asterisk forcefully will solve the issue. We're running Asterisk 1.4.18.
Thanks for any help!
[parallelparking]
exten = _7[89]X,1,Noop(Attempting to parallel park...)
exten = _7[89]X,n,Answer
exten = _7[89]X,n,Set(PARKINGEXTEN=${EXTEN})
exten = _7[89]X,n,GotoIf($[${BLINDTRANSFER} != ]?dopark:dounpark)
exten = _7[89]X,n(dopark),Noop(Going to try to park this call)
exten = _7[89]X,n,Set(RECALLEXTEN=${BLINDTRANSFER:4:4})
exten =
_7[89]X,n,ParkAndAnnounce(PARKED|180|Local/parkedannou...@parallelparking|parkreturn,${RECALLEXTEN},1)
exten = _7[89]X,n,Hangup
exten = _7[89]X,n(dounpark),Noop(Going to try to un-park this call)
exten = _7[89]X,n,ParkedCall(${EXTEN})
exten = _7[89]X,n,Hangup
exten = parkedannounce,1,Noop
exten = parkedannounce,n,Answer
exten = parkedannounce,n,Wait(1)
exten = parkedannounce,n,Hangup
[parkreturn]
exten = _,1,Noop(Returning Parked Call)
exten = _,n,SIPAddHeader(Alert-Info: info=${AASTRA_PARKRINGBACK})
exten = _,n,Set(CALLERID(name)=FrPark:${CALLERID(name)})
exten = _,n,Dial(SIP/${EXTEN},60)
exten = _,n,Hangup
Jonathan Bailey
Marshall County, Iowa
1 E Main St, Marshalltown, IA 50158
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Re: [asterisk-users] Gosub behavior change =1.6.0.5 to 1.6.0.6
On Tue, 2009-02-24 at 16:58 -0600, Tilghman Lesher wrote: If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. While I personally believe it's a bug, it has been in Asterisk for a very long time, and I know from teaching Asterisk training classes that there are *many* *many* people abusing this in their dialplans. I'd be quite hesitant to change this behavior without some very large warning signs. -- Jared Smith Digium, Inc. | Training Manager ___ -- 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] Polycom Spectralink 8002 Configuration
Mark, Are you still having trouble with your 8002? I had a lot of trouble with mine initially, but after playing with it for about 8 hours I figured it out. Now it works great all around our office. Our NOC technician loves it! There is a problem with the sample configs that Polycom publishes. I started by un-commenting and modifying the portions that related to an Asterisk setup. However, that seemed to be the source of my problem in the end. I don't know if the phone simply can't parse the length of the sample file, or if there are some errors in the sample file that I missed. As soon as I trimmed the config file down to just the necessary components, the phone started to work! Bob On Mon, 2009-02-23 at 21:07 -0500, M Hulber wrote: I have a new Polycom Spectralink 8002 and am having trouble with the configuration or the unit but I can't see what's wrong. The unit does not seem to even attempt to register with the Asterisk proxy but I can make calls to it. I have viewed the syslog from the device which it will actually write to the asterisk server so I know it can be reached. I have also run a sip debug and see no registration traffic from the unit. It also pulls the configs from the tftp server on the asterisk box ok. Does anyone have a sample set of configs that work? I have samples for the Polycom side but haven't seen the match on the asterisk side. Since I don't even see traffic, I can't think that it's even an authentication issue. When I dial from the device it just sits there, basically. MARK. -- sip_allusers.cfg: (I've tried most variations on theses settings) ## FOR PROXY1_TYPE = ASTERISK #PROXY1_ADDR = 192.168.2.80:5060# replace the ip address with the Asterisk Server's Address PROXY1_ADDR = 192.168.2.80 # replace the ip address with the Asterisk Server's Address PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = enable PROXY1_HOLD_IP0 = disable PROXY1_PRACK = enable #PROXY1_REREG_SECS=3600 PROXY1_REREG_SECS=35 PROXY1_KEEPALIVE_SECS=14 #PROXY1_DOMAIN = asterisk# Replace this with your SIP Domain's name PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = 864 # Put Your Voice Mail Sytem's Pilot Number here sip_2000.cfg: LINE1 = 2000 LINE1_PROXY = 1 LINE1_CALLID = 2000 #LINE1_AUTH= 2000; 2000 sip.conf: ; Polycom Spectralink 8002 [2000] type=friend host=192.168.3.123 ;port=5060 secret=2000 username=2000 ;fromuser=2000 ;authuser=2000 qualify=no ; turned this off to stop asterisk side initiated traffic context=spectra_default dtmfmode=rfc2833 disallow=all allow=ulaw mailbox...@default canreinvite=yes callgroup=1 pickupgroup=1 accountcode=Home nat=no Syslog: Feb 23 20:25:06 192.168.3.123 Jan 1 00:18:24.57 0090.7a0a.13f3 (192.168.003.123) [0007] Call start, AP 0014.d1c2.70fe (-32 dBm) Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.87 0090.7a0a.13f3 (192.168.003.123) [0008] Number Abufs: 26 Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.87 0090.7a0a.13f3 (192.168.003.123) [0009] Number Fbufs: 2 Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.88 0090.7a0a.13f3 (192.168.003.123) [000a] Max Number Abufs: 359 Feb 23 20:25:09 192.168.3.123 Jan 1 00:18:26.88 0090.7a0a.13f3 (192.168.003.123) [000b] Max Number Fbufs: 33 Feb 23 20:25:11 192.168.3.123 Jan 1 00:18:29.57 0090.7a0a.13f3 (192.168.003.123) [000c] NStat: 0014.d1c2.70fe (-30 dBm), Tx 3704, Rx 43841, BTx 2, BRx 2766, MTx 0, MRx 0, Tx Drop 3 (0.1%), Tx Retry 96 (2.7%), Rx Retry 19 (0.0%) Feb 23 20:25:16 192.168.3.123 Jan 1 00:18:33.87 0090.7a0a.13f3 (192.168.003.123) [000d] Number Abufs: 46 Feb 23 20:25:16 192.168.3.123 Jan 1 00:18:33.87 0090.7a0a.13f3 (192.168.003.123) [000e] Number Fbufs: 3 Feb 23 20:25:16 192.168.3.123 Jan 1 00:18:34.57 0090.7a0a.13f3 (192.168.003.123) [000f] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3707, Rx 43996, BTx 2, BRx 2773, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96 (0.0%), Rx Retry 19 (0.0%) Feb 23 20:25:21 192.168.3.123 Jan 1 00:18:39.57 0090.7a0a.13f3 (192.168.003.123) [0010] NStat: 0014.d1c2.70fe (-36 dBm), Tx 3708, Rx 44284, BTx 2, BRx 2792, MTx 0, MRx 0, Tx Drop 3 (0.0%), Tx Retry 96 (0.0%), Rx Retry 19 (0.0%) Feb 23 20:25:26 192.168.3.123 Jan 1 00:18:44.36 0090.7a0a.13f3 (192.168.003.123) [0011] Call end, AP 0014.d1c2.70fe (-36 dBm) ___ -- 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
[asterisk-users] TE121 on Asterisk
Hello, I just bought a TE121 T1/E1 card, and now trying to install it on a 1.4.23.1 asterisk with dahdi 2.1.0.4 Actually first everything went on well and i managed to see my card on dahdi. Here's the output: #asterisk# dahdi_hardware pci::04:08.0 wcte12xp+d161:8000 Wildcard TE121 and this is the scan: -- asterisk# dahdi_scan [1] active=yes alarms=RED description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VPMADT032 location=PCI Bus 04 Slot 09 basechan=1 totchans=24 irq=17 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF -- and this is the proc output asterisk# cat /proc/dahdi/1 Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) B8ZS/ESF RED IRQ misses: 1 1 WCT1/0/1 Clear RED (EC: MG2) 2 WCT1/0/2 Clear RED (EC: MG2) 3 WCT1/0/3 Clear RED (EC: MG2) 4 WCT1/0/4 Clear RED (EC: MG2) 5 WCT1/0/5 Clear RED (EC: MG2) 6 WCT1/0/6 Clear RED (EC: MG2) 7 WCT1/0/7 Clear RED (EC: MG2) 8 WCT1/0/8 Clear RED (EC: MG2) 9 WCT1/0/9 Clear RED (EC: MG2) 10 WCT1/0/10 Clear RED (EC: MG2) 11 WCT1/0/11 Clear RED (EC: MG2) 12 WCT1/0/12 Clear RED (EC: MG2) 13 WCT1/0/13 Clear RED (EC: MG2) 14 WCT1/0/14 Clear RED (EC: MG2) 15 WCT1/0/15 Clear RED (EC: MG2) 16 WCT1/0/16 Clear RED (EC: MG2) 17 WCT1/0/17 Clear RED (EC: MG2) 18 WCT1/0/18 Clear RED (EC: MG2) 19 WCT1/0/19 Clear RED (EC: MG2) 20 WCT1/0/20 Clear RED (EC: MG2) 21 WCT1/0/21 Clear RED (EC: MG2) 22 WCT1/0/22 Clear RED (EC: MG2) 23 WCT1/0/23 Clear RED (EC: MG2) 24 WCT1/0/24 HDLCFCS RED - I want to change it to E1 instead of T1. here comes the problem. Default system.conf under /etc/dundi is: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 echocanceller=mg2,1-23 loadzone= us defaultzone = us -- to make it work as E1, if i write a new span like span=1,1,0,ccs,hdb3,crc4 i got the following error when i type dahdi_cfg dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels to configure. DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) How can i set a working E1. Any examples will welcome.. ___ -- 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] TE121 on Asterisk
E-1s are 30 channels with D-Channel on 16. to make it work as E1, if i write a new span like span=1,1,0,ccs,hdb3,crc4 i got the following error when i type dahdi_cfg dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: D-channel (Default) (Slaves: 24) 24 channels to configure. DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) How can i set a working E1. Any examples will welcome.. ___ -- 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 -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.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] Polycom Spectralink 8002 Configuration
On Tue, 24 Feb 2009, Michael Graves wrote: It seems to me that based upon your comments you miss the point of the product. It's design targets large commercial concerns, school campuses, corporate parks, etc...not making free calls from Starbucks. Completely right. I assumed it was a generic wifi based SIP phone. I had one under test for several months and it behaved really well on my WLAN using a Netgear comsumer N type rouiter/AP with WMM. WMM is essentially a wireless QoS mechanism. Without it you cannot assure voice quality if there's anything else using the WLAN. Granted, the phone is a bit fiddly to provision. In it's intended target markets that's not a problem. If you want to make free calls from hotspots you're far better of with trashy consumer oriented stuff that has a built-in web browser. In many cases you need it to authenticate against the hotspot. The best option seems to be a SIP client on a dual mode cell phone. But then, why use the wifi when you have a cell phone in your hand? Minutes are cheap in either case. Because I still have this dream of having my extension in my hand. I've had very poor luck with my iPhone and SIP clients I have tried. The best I have been able to manage is X-Lite on my laptop, which actually works very well. My laptop doesn't fit in my pocket, though, sadly :) There does seem to be a market, if small, for a wifi enabled SIP phone that maybe isn't a full fledged cell phone. Although I can see how the Polycom phone might be useful in a wide campus environment where it may roam among many wifi nodes, that seems a pretty small market segment. For a regular office or building a DECT phone plugged into an ATA seems to be the way to go. The Polycom phone, totally against the norm for Polycom IMO, looks and feels cheap and has funky buttons :) I actually haven't gotten mine to work at all. Mind pasting the config that works for you? Just around the house here I am using DD-WRT on a Linksys WRT54G, which does support WMM. Cheers, j ___ -- 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] Polycom Spectralink 8002 Configuration
On Wed, 25 Feb 2009, Bob Pierce wrote: Mark, Are you still having trouble with your 8002? I had a lot of trouble with mine initially, but after playing with it for about 8 hours I figured it out. Now it works great all around our office. Our NOC technician loves it! There is a problem with the sample configs that Polycom publishes. I started by un-commenting and modifying the portions that related to an Asterisk setup. However, that seemed to be the source of my problem in the end. I don't know if the phone simply can't parse the length of the sample file, or if there are some errors in the sample file that I missed. As soon as I trimmed the config file down to just the necessary components, the phone started to work! Aha! Mind posting that config? Cheers, j ___ -- 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] TE121 on Asterisk
Oguzhan Kayhan wrote: I want to change it to E1 instead of T1. here comes the problem. If it's anything like the older cards, there is a jumper on the card that sets it to T1/E1 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jared Smith wrote: While I personally believe it's a bug, it has been in Asterisk for a very long time, and I know from teaching Asterisk training classes that there are *many* *many* people abusing this in their dialplans. I'd be quite hesitant to change this behavior without some very large warning signs. I think that the appropriate time is during an upgrade to a new version. Even from 1.6.0 to 1.6.1 would be okay, given that the behavior change is documented in the upgrade.txt document. Doing it from a .05 to a .06 release can certainly catch many off-guard. BK -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFJpWAfCFu3bIiwtTARAp1AAJoDgKg1o0UPHg/0uGXesOVMZyP+0wCfXzbY XWUUOuxPwKdWG2xsbEGV2PY= =6+mm -END PGP SIGNATURE- ___ -- 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] TE121 on Asterisk
On Wed, 2009-02-25 at 17:00 +0200, Oguzhan Kayhan wrote: I want to change it to E1 instead of T1. To change it from T1 mode to E1 mode, you need to move the jumper on the card. (If you don't have physical access to the card, you can also override the jumper with a parameter to the kernel module.) -- Jared Smith Digium, Inc. | Training Manager ___ -- 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] dahdi wcb4xxp and fax
Hi all, I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Our setup uses a digium b410p card with asterisk 1.6, latest libpri and dahdi, hylafax with iaxmodem, and all this on 1 machine. chan_dahdi.conf contains: faxdetect=both When receiving a fax call, hylafax (iaxmodem) answers the call after the obligatory wait of 3 seconds (fax detection) but to me it seems that echo cancellation is still being done. Any pointers on this or workarounds? We're back to our old misdn setup for now ;) Here's some output from dahdi show channel 1 (the one that had the fax connection going), i cut out some non-related stuff : *CLI dahdi show channel 4 Signalling Type: ISDN BRI Point to Point Owner: DAHDI/4-1 Real: DAHDI/4-1 Callwait: None Threeway: None Confno: -1 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: yes Pulse phone: no DND: no Echo Cancellation: 128 taps (unless TDM bridged) currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Regards, stoffell ___ -- 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] TE121 on Asterisk
E-1s are 30 channels with D-Channel on 16. Ok, so i replaced the channels as D-chan 16 And now i get the following error. This card is suppose to be both e1-t1 as i understand...Or did i receive a card with only T1 support?? How will i configure it to work with e1? dahdi_cfg -vvv DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: D-channel (Default) (Echo Canceler: mg2) (Slaves: 16) Channel 17: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 17) Channel 18: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 18) Channel 19: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 19) Channel 20: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 20) Channel 21: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 21) Channel 22: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 22) Channel 23: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) 30 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 Setting echocan for channel 9 to mg2 Setting echocan for channel 10 to mg2 Setting echocan for channel 11 to mg2 Setting echocan for channel 12 to mg2 Setting echocan for channel 13 to mg2 Setting echocan for channel 14 to mg2 Setting echocan for channel 15 to mg2 Changing signalling on channel 16 from Clear channel to HDLC with FCS check Setting echocan for channel 16 to mg2 Setting echocan for channel 17 to mg2 Setting echocan for channel 18 to mg2 Setting echocan for channel 19 to mg2 Setting echocan for channel 20 to mg2 Setting echocan for channel 21 to mg2 Setting echocan for channel 22 to mg2 Setting echocan for channel 23 to mg2 Changing signalling on channel 24 from HDLC with FCS check to Clear channel DAHDI_CHANCONFIG failed on channel 25: No such device or address (6) to make it work as E1, if i write a new span like span=1,1,0,ccs,hdb3,crc4 i got the following error when i type dahdi_cfg dahdi_cfg - DAHDI Tools Version - 2.1.0.2 DAHDI Version: 2.1.0.4 Echo Canceller(s): MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 03: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 03) Channel 04: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 04) Channel 05: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 05) Channel 06: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 06) Channel 07: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 07) Channel 08: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 08) Channel 09: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 09) Channel 10: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 10) Channel 11: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 11) Channel 12: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 12) Channel 13: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 13) Channel 14: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 14) Channel 15: Clear channel (Default) (Echo Canceler: mg2) (Slaves: 15) Channel 16: Clear channel (Default) (Echo
Re: [asterisk-users] TE121 on Asterisk
Oguzhan Kayhan wrote: I want to change it to E1 instead of T1. here comes the problem. If it's anything like the older cards, there is a jumper on the card that sets it to T1/E1 Doug Yes, I just noticed the jumper on the card. Thanks a lot. -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] SIP_CODEC variable
2009/2/25 Jeff LaCoursiere j...@jeff.net
On Wed, 25 Feb 2009, Jared Smith wrote:
On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate Page command call. But I get this in th CLI:
NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
${SIP_CODEC} variable because it is not shared by both ends.
This is a wild guess (and I don't currently have the time to check it
out properly), but if my memory serves me the Polycom phones don't
support the GSM codec. You might try ulaw instead.
True, that. They do G.729 though!
j
If my memory serves me right, there is an opened bug in Mantis about
SIP_CODEC not being presently applied to both legs of a call.
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Re: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
It all has to do with interconnection agreements with the ILEC and if the reseller has numbering resources in the requested area. Looks like BroadVoice does have all those elements taken care of. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 3:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide us VOIP services in the 916 area code. I am wondering how can I get the bandwidth.com service, Thanks, Vikas ___ -- 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] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA
Frank Bulk wrote: It all has to do with interconnection agreements with the ILEC and if the reseller has numbering resources in the requested area. Looks like BroadVoice does have all those elements taken care of. why not just lie about your address ? that would seem like the obvious solution. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 3:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] bandwidth.com will not sell me a sip line since the address is in Citrus Heights CA I called bandwidth.com to buy a sip line from them for $30 a month. But they said they will not sell me a sip line since the address on the account is Citrus Heights CA and they can not provide services in that area. On asking further the person clarified that there is no e911 service available in the 916 area code for bandwidth.com But other providers like www.broadvoice.com are able to provide us VOIP services in the 916 area code. I am wondering how can I get the bandwidth.com service, Thanks, Vikas ___ -- 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 ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. regards klaus ___ -- 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] dahdi wcb4xxp and fax
stoffell wrote: I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Make sure that you're using the latest mISDN drivers. Thanks, Lee. ___ -- 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] Polycom Spectralink 8002 Configuration
On Wed, 2009-02-25 at 15:13 +, Jeff LaCoursiere wrote: Aha! Mind posting that config? My sip_allusers.cfg looks like this: CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.8.1:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 PROXY1_DOMAIN = 192.168.8.1 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 My sip_.cfg looks like this: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = NOC Tech LINE1_AUTH= ; secret LINE2 = LINE2_PROXY = 1 LINE2_CALLID = NOC Tech LINE2_AUTH= ; secret Bob ___ -- 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] Polycom Spectralink 8002 Configuration
I agree with the comments on the intended target market for this phone. In defense of Polycom, if your TFTP server is external you could connect to a remote access point by setting up WEP/WPA fairly easily from Starbucks or wherever you are. If it requires web authentication to get through the firewall then I suppose you would have a problem. Your config files would need to be location agnostic but that's not such a big deal. This is the only WIFI phone I have come across that has decent reliability reviews and a fairly reasonable price point. Having had it for a couple days now, it is very simple for the user (not necessarily the admin). It appears that not all APs explicitly advertise in their specifications that they support WMM. I have an AP that supports WISH but nowhere do I see any documentation that it supports WMM but it works ok. I think WISH leverages WMM from the brief searching I did. Jeff LaCoursiere wrote: On Tue, 24 Feb 2009, Michael Graves wrote: It seems to me that based upon your comments you miss the point of the product. It's design targets large commercial concerns, school campuses, corporate parks, etc...not making free calls from Starbucks. Completely right. I assumed it was a generic wifi based SIP phone. I had one under test for several months and it behaved really well on my WLAN using a Netgear comsumer N type rouiter/AP with WMM. WMM is essentially a wireless QoS mechanism. Without it you cannot assure voice quality if there's anything else using the WLAN. Granted, the phone is a bit fiddly to provision. In it's intended target markets that's not a problem. If you want to make free calls from hotspots you're far better of with trashy consumer oriented stuff that has a built-in web browser. In many cases you need it to authenticate against the hotspot. The best option seems to be a SIP client on a dual mode cell phone. But then, why use the wifi when you have a cell phone in your hand? Minutes are cheap in either case. Because I still have this dream of having my extension in my hand. I've had very poor luck with my iPhone and SIP clients I have tried. The best I have been able to manage is X-Lite on my laptop, which actually works very well. My laptop doesn't fit in my pocket, though, sadly :) There does seem to be a market, if small, for a wifi enabled SIP phone that maybe isn't a full fledged cell phone. Although I can see how the Polycom phone might be useful in a wide campus environment where it may roam among many wifi nodes, that seems a pretty small market segment. For a regular office or building a DECT phone plugged into an ATA seems to be the way to go. The Polycom phone, totally against the norm for Polycom IMO, looks and feels cheap and has funky buttons :) I actually haven't gotten mine to work at all. Mind pasting the config that works for you? Just around the house here I am using DD-WRT on a Linksys WRT54G, which does support WMM. Cheers, j ___ -- 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] dahdi wcb4xxp and fax
Lee Howard wrote: stoffell wrote: I wanted to switch from my current setup (mISDN) to the native dahdi with b410p support (wcb4xp). All works fine for normal phone calls but not for faxing. Faxes are distorted, if arriving at all, and hylafax logs the usual bad stuff (HDLC frame not byte-oriented.) Make sure that you're using the latest mISDN drivers. Even the latest mISDN gives variable results. Some people say its OK. Some people say its hopeless. It probably varies with the machine its running in. 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] dahdi wcb4xxp and fax
On Wed, Feb 25, 2009 at 5:28 PM, Steve Underwood ste...@coppice.org wrote: Lee Howard wrote: Make sure that you're using the latest mISDN drivers. Even the latest mISDN gives variable results. Some people say its OK. Some people say its hopeless. It probably varies with the machine its running in. the whole point is I wanted to move away from mISDN (for other reasons) to the digium-way so I can use native digium (and only digium) software. :-) regards, stoffell ___ -- 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] Polycom Spectralink 8002 Configuration
Bob, Ok, that's the route I ended up taking where all lines are the same user. I put the AUTH an LINEn_AUTH in the phone instead. I wanted to be able to set up so that each line is a different peer like below: sip_.cfg: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = ABC Tech LINE1_AUTH= ; secret LINE2 = LINE2_PROXY = 1 LINE2_CALLID = ABC Sales LINE2_AUTH= ; secret So I'm thinking, would this work if I had a sip_.conf as well as a sip_.conf? What the relationship between the LINEs in the sip_.cfg and the Reg on the phone? What's the relationship between the AUTH and the LINEn_AUTH? This is just a bit confusing to me. Basically, I want to treat the phone as a multiple extension phone instead of a single user phone. Where each extension (LINE) represents itself as a unique peer when communicating with Asterisk and is registered uniquely. Bob Pierce wrote: On Wed, 2009-02-25 at 15:13 +, Jeff LaCoursiere wrote: Aha! Mind posting that config? My sip_allusers.cfg looks like this: CODECS = g711u, g711a PROXY1_TYPE = Asterisk PROXY1_ADDR = 192.168.8.1:5060 #PROXY1_KEYPRESS_2833 = enable PROXY1_KEYPRESS_INFO = disable PROXY1_HOLD_IP0 = disable #PROXY1_PRACK = enable PROXY1_REREG_SECS=3600 PROXY1_KEEPALIVE_SECS=14 PROXY1_DOMAIN = 192.168.8.1 PROXY1_CALLID_PER_LINE = disable PROXY1_MAIL_ACCESS = *97 My sip_.cfg looks like this: AUTH = ; secret LINE1 = 1234 LINE1_PROXY = 1 LINE1_CALLID = ABC Tech LINE1_AUTH= 1234; secret LINE2 = 1234 LINE2_PROXY = 1 LINE2_CALLID = ABC Tech LINE2_AUTH= 1234; secret Bob ___ -- 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] SIP_CODEC variable
Thanks, I took it for granted that the phones did support gsm...silly me.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jared Smith
Sent: Wednesday, February 25, 2009 9:15
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP_CODEC variable
On Wed, 2009-02-25 at 07:54 -0500, Mike wrote:
I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the
appropriate Page command call. But I get this in th CLI:
NOTICE[4764]: chan_sip.c:3706 try_suggested_sip_codec: Ignoring
${SIP_CODEC} variable because it is not shared by both ends.
This is a wild guess (and I don't currently have the time to check it
out properly), but if my memory serves me the Polycom phones don't
support the GSM codec. You might try ulaw instead.
--
Jared Smith
Digium, Inc. | Training Manager
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Re: [asterisk-users] Aastra phones
One thing which I can't figure out, although it certainly looks simple, is to update the firmware though FTP (not TFTP). I have set the ftp provisioning server in the Aastra phone, and put the firmware file 9143i.st in the root folder where the login/password pair ends up. Everything is entered correctly, or so it seems (works fine with my Polycoms). I believe that the older firmware for the Aastra phone will only update from TFTP. I am not sure what rev level this changed at though. JohnM John, You are absolutely correct, I did a two step upgrade to test this (from 2.0.5 to 2.4.0 to 2.4.1). The first update only worked through TFTP, but the second one worked with FTP. Thanks for the tip, Mike ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. -- Tilghman ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher wrote: On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. I think the point being made was that a lot of people thought this was a feature, not a bug. I assume you're asserting the the dev's did not expect this behaviour, even if a large group of users did. That's OK. But there's still the question about why this behaviour is so bad/inconsistent/something that it should be changed. Simply labeling it a bug is just a conclusion. Why is it a bug??? sean ___ -- 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] Polycom Spectralink 8002 Configuration
On Wed, 2009-02-25 at 11:37 -0500, M Hulber wrote: So I'm thinking, would this work if I had a sip_.conf as well as a sip_.conf? What the relationship between the LINEs in the sip_.cfg and the Reg on the phone? What's the relationship between the AUTH and the LINEn_AUTH? This is just a bit confusing to me. Basically, I want to treat the phone as a multiple extension phone instead of a single user phone. Where each extension (LINE) represents itself as a unique peer when communicating with Asterisk and is registered uniquely. OK, so the confusing thing that was not documented by Polycom is this: At the bottom of page 46, the grey box mentions that each handset needs a config file (which I expected), but it does not clearly state why you would name them sip_JohnDoe.cfg or sip_3001.cfg - This was a little counter intuitive for me until I realized it was related to a username that was entered in the phone's menu. So if you enter the user on the phone as it will pick up the sip_.cfg file and if you enter on the phone it will pick up the sip_.cfg file. I think you would want to break your config out into two files like this: sip_.cfg: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = ABC Tech LINE1_AUTH= ; secret LINE2 = 5 LINE2_PROXY = 1 LINE2_CALLID = ABC Tech LINE2_AUTH= ; secret sip_.cfg: AUTH = ; secret LINE1 = LINE1_PROXY = 1 LINE1_CALLID = ABC Sales LINE1_AUTH= ; secret LINE2 = LINE2_PROXY = 1 LINE2_CALLID = ABC Sales LINE2_AUTH= ; secret Or, you could leave it like you have and have the phone register to both extensions at the same time. I'm not sure what you should do with the first line in that case. Bob ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher schrieb: On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. voip-info.org is also beeing known as where to find documentation where Asterisk itself lacks of documentation. The problem here is not voip-info, but that the old behavior is there and used since at least Nov 2005. Changing an over 3 years old behavior is not nice. regards klaus ___ -- 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] Multiple SIPGate accounts.
Thanks Klaus. Putting both in the same context solved my issue! ___ -- 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] cannot allocate memory
Hi: i have a hosted server with asterisk and a2billing as a billing plattform, when i am trying to enter the server remotely by ssh, memory error message displayed: -bash: fork: Cannot allocate memory i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the system is'nt 1GB of RAM sufficient for this volume of calls on Asterisk. and when iam using top command this is what i get: top - 20:20:25 up 2:15, 1 user, load average: 0.57, 0.22, 0.13 Tasks: 100 total, 1 running, 36 sleeping, 0 stopped, 63 zombie Cpu(s): 7.7% us, 2.3% sy, 0.0% ni, 90.0% id, 0.0% wa, 0.0% hi, 0.0% si, Mem: 1048576k total, 316088k used, 732488k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached As i see it that the free memory is 732488k ,so it should'nt make this error. _ Windows Live™ Hotmail®…more than just e-mail. http://windowslive.com/howitworks?ocid=TXT_TAGLM_WL_t2_hm_justgotbetter_howitworks_022009___ -- 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] DID's in a specific rate center
I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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] DID's in a specific rate center
If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 12:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID's in a specific rate center I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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
[asterisk-users] CDR - Asterisk-Stat and PHP5
Hi all, I don't know if its the right place to ask, but... Does any one have the asterisk-stat-v2 running with PHP5? Tks! -- Tiago Durante ,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,.,., Perseverance is the hard work you do after you get tired of doing the hard work you already did. -- Newt Gingrich ___ -- 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] cannot allocate memory
On Wed, 25 Feb 2009, wassim Darwish wrote: i have a hosted server with asterisk and a2billing as a billing plattform, when i am trying to enter the server remotely by ssh, memory error message displayed: -bash: fork: Cannot allocate memory This is not an Asterisk error message. i have 1GB RAM on the system ,and there is 15 to 25 concurrent calls on the system is'nt 1GB of RAM sufficient for this volume of calls on Asterisk. Yes. Way more than sufficient. and when iam using top command this is what i get: top - 20:20:25 up 2:15, 1 user, load average: 0.57, 0.22, 0.13 Tasks: 100 total, 1 running, 36 sleeping, 0 stopped, 63 zombie Cpu(s): 7.7% us, 2.3% sy, 0.0% ni, 90.0% id, 0.0% wa, 0.0% hi, 0.0% si, Mem: 1048576k total, 316088k used, 732488k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached As i see it that the free memory is 732488k ,so it should'nt make this error. Personally, I'm not a big fan of zombies. If this is what top displays while you are getting the error message on another shell, it is not a free memory issue. Maybe some other resource like file handles is being sucked up by your zombies and bash is misreporting the error. Does the error fix itself or do you need to reboot the box? Unrelated, but I would add a swap file just in case you need it at some point. While swapping is a somewhat bad thing, I prefer it to failing or locked processes. Also, you didn't say what OS, OS version or Asterisk version you are running. Updating the OS (for CentOS, sudo yum update) and Asterisk may resolve your problem. If not, you will have a better chance finding a solution if you are running something reasonably current. (Says he who still runs 1.2...) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] DID's in a specific rate center
you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). But if I spoof the DID the person receiving the call will not be able to get back to me. So I do not think that is going to work for me. Vikas On Wed, Feb 25, 2009 at 12:56 PM, Danny Nicholas da...@debsinc.com wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 12:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID's in a specific rate center I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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 ___ -- 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] DID's in a specific rate center
Danny Nicholas wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. You do know that that's illegal, right? ___ -- 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] DID's in a specific rate center
Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk Sent: Wednesday, February 25, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Danny Nicholas wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. You do know that that's illegal, right? ___ -- 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] DID's in a specific rate center
If you have 1 real DID that you spoof from, the user will call back the real DID. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). But if I spoof the DID the person receiving the call will not be able to get back to me. So I do not think that is going to work for me. Vikas On Wed, Feb 25, 2009 at 12:56 PM, Danny Nicholas da...@debsinc.com wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 12:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DID's in a specific rate center I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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 ___ -- 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
[asterisk-users] Patton 5.3. How to get incoming calls ?
Hi, I'm trying to configure a 4638 to pass inbound and outbound to and from ISDN and SIP interfaces. I'm using web interface at the moment. Setup is: ISDN -- BRI -- Patton 4638 -- SIP Asterisk -- SIP -- IP Phone I can call from IP phone but can't receive any incoming call : I can't see any SIP message coming in when a call comes in. Previously, with 4.2 firmware, you just have to edit routing table binding ISDN ports to SIP interface to get calls coming in but now with 5.3, configuration process changed. Here is an extract from my running config. Any idea ? Regards context cs switch routing-table called-e164 appels_provenance_ISDN route [0-9]+ dest-service ASTERISK_SRV route default dest-service ASTERISK_SRV routing-table called-uri appels_vers_ISDN route default dest-service isdnports mapping-table called-e164 to called-ip transfo map [0-9]+ to 192.168.100.254 mapping-table called-e164 to called-uri transfo2 interface isdn IF-PBX route call dest-table appels_provenance_ISDN interface isdn IF-PBX2 route call dest-table appels_provenance_ISDN interface isdn IF-PBX3 route call dest-table appels_provenance_ISDN interface isdn IF-PBX4 route call dest-table appels_provenance_ISDN interface sip IF-ASTERISK bind context sip-gateway ASTERISK route call dest-table appels_vers_ISDN service sip-location-service ASTERISK_SRV bind location-service ASTERISK_SRV mode hunt hunt-timeout 20 service hunt-group isdnports drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF-PBX route call 2 dest-interface IF-PBX2 route call 3 dest-interface IF-PBX3 context cs switch no shutdown authentication-service patton realm 1 asterisk username patton password Otx2vJCEWP+8Bb6tqoGkwA== encrypted location-service ASTERISK_SRV domain 1 192.168.100.254 5060 domain 2 asterisk 5060 identity-group default identity patton alias name patton authentication outbound authenticate 1 authentication-service patton username patton registration outbound registrar 192.168.100.254 5060 proxy none lifetime 3600 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound use profile tone-set default use profile voip default use profile sip default preferred-transport-protocol udp invite-transaction-timeout 32 non-invite-transaction-timeout 32 call inbound use profile tone-set default use profile voip default use profile sip default ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
On Wednesday 25 February 2009 11:19:08 sean darcy wrote: Tilghman Lesher wrote: On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. I think the point being made was that a lot of people thought this was a feature, not a bug. I assume you're asserting the the dev's did not expect this behaviour, even if a large group of users did. That's OK. But there's still the question about why this behaviour is so bad/inconsistent/something that it should be changed. Simply labeling it a bug is just a conclusion. Why is it a bug??? It's a bug, because the i extension has a very limited intended usage, and any additional cases where the i extension is implicitly invoked is therefore a bug. This thread has convinced me not to change Goto in 1.6.0, but I absolutely defend fixing this bug in Gosub, given that I'm the designer of it, and it was never supposed to fail into the i extension. -- Tilghman ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Why not expand the usage of the i extension? If not in 1.6.0, then some later 1.6. Call it a feature enhancement. Tilghman Lesher wrote: On Wednesday 25 February 2009 11:19:08 sean darcy wrote: Tilghman Lesher wrote: On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote: Tilghman Lesher schrieb: On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote: Barry L. Kline wrote: that is supposed to gosub into the incoming extension at priority 1. Versions before 1.6.0.6 would drop into the incoming,i,1 priority if the requested extension wasn't present in the incoming context. Really strange that Goto and Gosub behave different. If Goto behaves that way, that's a bug. As stated in a prior email, the i extension should only be implicitly invoked when waiting for a new extension and the typed extension does not match anything. FYI: If you take a look at the history of http://www.voip-info.org/wiki/index.php?page=Asterisk+i+extension you will find out that the old behavior is there since at least Nov. 2005, and probably used since then. voip-info.org is best known for being often wrong. I think the point being made was that a lot of people thought this was a feature, not a bug. I assume you're asserting the the dev's did not expect this behaviour, even if a large group of users did. That's OK. But there's still the question about why this behaviour is so bad/inconsistent/something that it should be changed. Simply labeling it a bug is just a conclusion. Why is it a bug??? It's a bug, because the i extension has a very limited intended usage, and any additional cases where the i extension is implicitly invoked is therefore a bug. This thread has convinced me not to change Goto in 1.6.0, but I absolutely defend fixing this bug in Gosub, given that I'm the designer of it, and it was never supposed to fail into the i extension. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.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] DID's in a specific rate center
http://en.wikipedia.org/wiki/Caller_ID_spoofing Danny Nicholas wrote: Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk Sent: Wednesday, February 25, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Danny Nicholas wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. You do know that that's illegal, right? ___ -- 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 ___ -- 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] Call from '6000' to extension rejected because extension not found
Call from '6000' to extension 'xx' rejected because extension not found. ___ -- 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] DID's in a specific rate center
So they are going to (eventually) make a legitimate (in some cases) practice Illegal because of spammers. Another blow for Libertarianism in the U.S. ! Don't know how this effects overseas readers. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonn Taylor Sent: Wednesday, February 25, 2009 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center http://en.wikipedia.org/wiki/Caller_ID_spoofing Danny Nicholas wrote: Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk Sent: Wednesday, February 25, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Danny Nicholas wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. You do know that that's illegal, right? ___ -- 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 ___ -- 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] Call from '6000' to extension rejected becauseextension not found
Dialplan problem, Chuck. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman Sent: Wednesday, February 25, 2009 2:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call from '6000' to extension rejected becauseextension not found Call from '6000' to extension 'xx' rejected because extension not found. ___ -- 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] Congestion Tone
Hello! Ive connected an avaya PABX with an asterisk box through h323, all calls from Avaya are sended to the asterisk. What I need is send to the AVAYA PABX a congestion tone when Zap channels are full. How I do it?Thanks for any idea! Cheers! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. ggonza...@despegar.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] DID's in a specific rate center
Spoofing the caller id is not an option for me. I am wondering how do I go about buying the DID's Thanks, Vikas On Wed, Feb 25, 2009 at 2:16 PM, Danny Nicholas da...@debsinc.com wrote: So they are going to (eventually) make a legitimate (in some cases) practice Illegal because of spammers. Another blow for Libertarianism in the U.S. ! Don’t know how this effects overseas readers. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonn Taylor Sent: Wednesday, February 25, 2009 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center http://en.wikipedia.org/wiki/Caller_ID_spoofing Danny Nicholas wrote: Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk Sent: Wednesday, February 25, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Danny Nicholas wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. You do know that that's illegal, right? ___ -- 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 ___ -- 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] DID's in a specific rate center
Just contact one of the providers mentioned in this forum, such as didvv.com, broadband.com or numerous others. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vikas Sent: Wednesday, February 25, 2009 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Spoofing the caller id is not an option for me. I am wondering how do I go about buying the DID's Thanks, Vikas On Wed, Feb 25, 2009 at 2:16 PM, Danny Nicholas da...@debsinc.com wrote: So they are going to (eventually) make a legitimate (in some cases) practice Illegal because of spammers. Another blow for Libertarianism in the U.S. ! Dont know how this effects overseas readers. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonn Taylor Sent: Wednesday, February 25, 2009 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center http://en.wikipedia.org/wiki/Caller_ID_spoofing Danny Nicholas wrote: Depends on the purpose. If I'm representing a client in another state with their permission, it's perfectly legit for me to spoof their number. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jay Milk Sent: Wednesday, February 25, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Danny Nicholas wrote: If you're using them outgoing only, you should consider spoofing the number (IE calling using XXX-XXX- and presenting as 916-854-). This would cost you $0.04-$0.06 per call, but you wouldn't need as many DID's. You do know that that's illegal, right? ___ -- 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 ___ -- 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 ___ -- 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] Realtime database function help
Hello Everyone! According to voip-info.org the correcy syntax for the realtime function is: REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read REALTIME(family|fieldmatch|value|field) on write It seems from the syntax that it is only possible to retrieve a full row according to the value of only of column. This translates in SQL language as Select * from family where fieldmath = value. Is there any way to have more control over the realtime function? Also, regarding the MYSQL function, I only saw documentation to query a database. Is there any way to update a database using that function? Thanks! Elliot ___ -- 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] SheevaPlug Development Kit
Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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] DID's in a specific rate center
Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can try contacting http://www.jnctn.com and ask if they can get numbers there. I do see they have others in the Sacramento area, in fact I have a Sacramento number with them already. If you don't and you just need outbound channels you can buy one (or more) DIDs and then use that as the caller-id setting for all the outbound calls. This is perfectly legal since you own the DID that you are using as the caller-id. The channels you are using for outbound calling don't have a DID associated with them so you need to associate it with one by setting the caller-id to an owned/valid DID. They don't have to be unique. What is illegal is to set caller-id to a fraudulent value such that the person on the other end will not be able to correctly identify the originator of the call. Vikas wrote: I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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] SheevaPlug Development Kit
please keep us informed about it. David 2009/2/25 Kristian Kielhofner kristian.kielhof...@gmail.com Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] SheevaPlug Development Kit
Yes please let us know how it works out. I have several projects in the works that this might work for. David fire wrote: please keep us informed about it. David 2009/2/25 Kristian Kielhofner kristian.kielhof...@gmail.com mailto:kristian.kielhof...@gmail.com Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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 -- Brent T. Vrieze CIM Automation Softare Engineer 507-216-0465 ___ -- 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] SheevaPlug Development Kit
Brent Vrieze wrote: Yes please let us know how it works out. I have several projects in the works that this might work for. sounds like a direct competitor of the nslu2's - The community following there is phenominal, but its nice to have some choice of platform as well. David fire wrote: please keep us informed about it. David 2009/2/25 Kristian Kielhofner kristian.kielhof...@gmail.com mailto:kristian.kielhof...@gmail.com Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] DID's in a specific rate center
Since it's not clear from this thread of conversation, do you need 100 unique DIDs? I apologize for not being more clear. I need 100 DID's. I already have channels which allow me to set the outgoing caller id. Depending on which extension is making the call I will be sending out the unique caller id. So that the person receiving the call can call back directly to the caller id that they received on their phone instead of going through the IVR hell. Vikas On Wed, Feb 25, 2009 at 3:13 PM, M Hulber asterisk-ad...@hulber.com wrote: Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can try contacting http://www.jnctn.com and ask if they can get numbers there. I do see they have others in the Sacramento area, in fact I have a Sacramento number with them already. If you don't and you just need outbound channels you can buy one (or more) DIDs and then use that as the caller-id setting for all the outbound calls. This is perfectly legal since you own the DID that you are using as the caller-id. The channels you are using for outbound calling don't have a DID associated with them so you need to associate it with one by setting the caller-id to an owned/valid DID. They don't have to be unique. What is illegal is to set caller-id to a fraudulent value such that the person on the other end will not be able to correctly identify the originator of the call. Vikas wrote: I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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 ___ -- 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] DID's in a specific rate center
Any idea what legal statues setting caller-id fraudulently falls under? Is there a federal law you can reference? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of M Hulber Sent: Wednesday, February 25, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DID's in a specific rate center Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can try contacting http://www.jnctn.com and ask if they can get numbers there. I do see they have others in the Sacramento area, in fact I have a Sacramento number with them already. If you don't and you just need outbound channels you can buy one (or more) DIDs and then use that as the caller-id setting for all the outbound calls. This is perfectly legal since you own the DID that you are using as the caller-id. The channels you are using for outbound calling don't have a DID associated with them so you need to associate it with one by setting the caller-id to an owned/valid DID. They don't have to be unique. What is illegal is to set caller-id to a fraudulent value such that the person on the other end will not be able to correctly identify the originator of the call. Vikas wrote: I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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 - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you. ___ -- 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] Call from '6000' to extension rejected because extension not found
Is this a question? Haha. Computer won't doesn't turn on. Got blck scrn. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chuck Coleman Sent: Wednesday, February 25, 2009 3:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call from '6000' to extension rejected because extension not found Call from '6000' to extension 'xx' rejected because extension not found. ___ -- 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] SheevaPlug Development Kit
On Wednesday 25 February 2009 14:59:02 Kristian Kielhofner wrote: Hello everyone, I just ordered one of these: http://www.marvell.com/products/embedded_processors/developer/kirkwood/shee vaplug.jsp Just over $110 with shipping but they are expecting the price to come down quite a bit: - 1.2Ghz ARM5 - 512MB RAM - Multiple flash storage options - Gigabit ethernet - USB 2.0 - 5 watt power usage They probably won't be shipping until late March but I thought I'd get my order in early. Of course one of my first tasks will be to get Asterisk running on it... ;) Looks like they finally fixed their shipping amounts. Yesterday, I was able to order one for $99 plus $0.00 next day shipping. -- Tilghman ___ -- 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] DID's in a specific rate center
Vikas, www.didforsale.com can get you the DIDs, please contact me off list. Jai Rangi jpra...@didforsale.com On Wed, Feb 25, 2009 at 1:35 PM, Vikas topg...@gmail.com wrote: Since it's not clear from this thread of conversation, do you need 100 unique DIDs? I apologize for not being more clear. I need 100 DID's. I already have channels which allow me to set the outgoing caller id. Depending on which extension is making the call I will be sending out the unique caller id. So that the person receiving the call can call back directly to the caller id that they received on their phone instead of going through the IVR hell. Vikas On Wed, Feb 25, 2009 at 3:13 PM, M Hulber asterisk-ad...@hulber.com wrote: Since it's not clear from this thread of conversation, do you need 100 unique DIDs? If you do: That NPA is owned by Pacbell with the central office: SCRMCA12 I don't know if anyone but Pacbell will have numbers in that NPA. Since I use them and am happy with the service, you can try contacting http://www.jnctn.com and ask if they can get numbers there. I do see they have others in the Sacramento area, in fact I have a Sacramento number with them already. If you don't and you just need outbound channels you can buy one (or more) DIDs and then use that as the caller-id setting for all the outbound calls. This is perfectly legal since you own the DID that you are using as the caller-id. The channels you are using for outbound calling don't have a DID associated with them so you need to associate it with one by setting the caller-id to an owned/valid DID. They don't have to be unique. What is illegal is to set caller-id to a fraudulent value such that the person on the other end will not be able to correctly identify the originator of the call. Vikas wrote: I need 100 DID's in a specific rate center (916-854-). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas ___ -- 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 ___ -- 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
[asterisk-users] AGI problem using mono (.Net)
Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly executes the program and moves to the next step of the dial plan, as though it didn't receive any commands from the program. We know it is running, and outputting its results, because we have called it from within a bash script, and in doing so, I set the script to output stdout to a txt file for testing (like this /var/log/asterisk/querylog). When we do this, the file does end up with the first line showing STREAM FILE filename. We're at a bit of a loss as to what's going on. We have checked filenames and are pretty sure that there are no typos and that the files are there. Further, I have a perl agi script using asterisk::agi that also does a STREAM FILE which runs without any problem. In our dial plan, my perl script runs, gets data from the user via the keypad, puts it in a channel variable, then exits, and his AGI script is immediately called as the next step of the dial plan receiving the channel variable as an argument. It seems that there are not as many out there using mono / .net with AGI. The few examples we've found online are a bit dated. Any help would be greatly appreciated. Thanks much! - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCP, Security+ Linux+, Network+, A+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 ___ -- 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] Gosub behavior change =1.6.0.5 to 1.6.0.6
Tilghman Lesher wrote: . ... but I absolutely defend fixing this bug in Gosub, given that I'm the designer of it, and it was never supposed to fail into the i extension. Wow. sean ___ -- 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] AGI problem using mono (.Net)
Douglas Mortensen wrote: I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly executes the program and moves to the next step of the dial plan, as though it didn't receive any commands from the program. We know it is running, and outputting its results, because we have called it from within a bash script, and in doing so, I set the script to output stdout to a txt file for testing (like this /var/log/asterisk/querylog). When we do this, the file does end up with the first line showing STREAM FILE filename. We're at a bit of a loss as to what's going on. We have checked filenames and are pretty sure that there are no typos and that the files are there. Further, I have a perl agi script using asterisk::agi that also does a STREAM FILE which runs without any problem. In our dial plan, my perl script runs, gets data from the user via the keypad, puts it in a channel variable, then exits, and his AGI script is immediately called as the next step of the dial plan receiving the channel variable as an argument. STDERR only goes to the Asterisk console if you are running 1.4 or later and enable agi debug in the CLI. I seem to recall something about AGIs not working correctly (streamfile or DTMF read) if your AGI script does not process the input Asterisk sends it on STDIN when Asterisk starts the AGI. I don't know if it applies here, but it's worth looking at. -- Eric Wieling * Asteria Solutions Group * Huntsville, AL Call centers * IVRs * Enterprise PBXs * Conferencing applications 256-705-0277 * http://www.asteriasgi.com/ * sa...@asteriasgi.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] AGI problem using mono (.Net)
On Wed, 25 Feb 2009, Douglas Mortensen wrote: I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly executes the program and moves to the next step of the dial plan, as though it didn't receive any commands from the program. Maybe you need a new developer? (Just kidding...) The agi debug command may shed some light on the problem. I'm not a big fan of AGIs outputting to STDERR. I like to pepper my AGIs with syslog() statements to show the program state and variables. We know it is running, and outputting its results, because we have called it from within a bash script, and in doing so, I set the script to output stdout to a txt file for testing (like this /var/log/asterisk/querylog). When we do this, the file does end up with the first line showing STREAM FILE filename. You can configure Asterisk to log a whole bunch of cruft to syslog with the following statement in logger.conf: syslog.local0 = debug,dtmf,error,event,info,notice,verbose,warning I'll apologize in advance if the text below underestimates your AGI skills. The AGI interface (is that redundant?) can be summarized as: 1) Asterisk sends a bunch of cruft (the AGI environment variables) to your program's STDIN. 2) Your program sends a request to Asterisk via STDOUT. 3) Asterisk sends a result to your program via STDIN. 4) Your program does something else. 5) go to step 2. It's very simple, but not very forgiving. Let's imagine a simple AGI that reads the ANI as a channel variable, parses out the area code and sets it as a channel variable named NPA. Thus, you can simulate the AGI environment with a shell script. For an example, imagine the following script named test-my-agi.sh: # the standard AGI environment variables echo agi_accountcode: echo agi_callerid: 1234567890 echo agi_calleridname: sedwards echo agi_callingani2: 0 echo agi_callingpres: 0 echo agi_callingtns: 0 echo agi_callington: 0 echo agi_channel: SIP/201-09456478 echo agi_context: newline echo agi_dnid: * echo agi_enhanced: 0.0 echo agi_extension: * echo agi_language: en echo agi_priority: 1 echo agi_rdnis: unknown echo agi_request: block-ani echo agi_type: SIP echo agi_uniqueid: 1195070681.28 echo # result for AGI command GET VARIABLE ANI echo 200 result=1 (5551234567) # result for AGI command SET VARIABLE NPA echo 200 result=1 # (end of test-my-agi.sh) You can test your agi by executing: ./test-my-agi.sh | my-agi Since your AGI requires specific interaction with Asterisk to play the file this method will not allow you to fully test it, but it may help identify where you are violating the protocol. This technique can even be used in an actual debugger like gdb so you can step through your code line by line. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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] Call from '6000' to extension rejected because extension not found
Please read this book: http://downloads.oreilly.com/books/9780596510480.pdf PaulH Chuck Coleman wrote: Call from '6000' to extension 'xx' rejected because extension not found. ___ -- 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] Realtime database function help
You can use the MYSQL function to just use an insert or update statement in
your dialplan. Look at my example below. Instead of using
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ callerid=${ARG1} and blockenabled = 1)
You could use:
exten = s,2,MYSQL(Query resultid ${connid} INSERT INTO\ callerid\
(callerid,blockenabled)\ VALUES\ ('${CALLERID(num)}', '1')\ )
I find that using the ODBC function works best for inserting data into the
MySQL databases.
Have a look at
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+func_odbc.
[globals]
realdb_host=hostnameformysqldb
realdb_user=mysqldbuser
realdb_pass=mysqldbpassword
realdb_db=mysqldbthatcontainsthevoicemailusers
[macro-checkblacklist]
; This Macro will check the blacklist table to see if the callerid of the
; caller exist and blockenabled =1 (TRUE). If the callerid is listed, then
; tell the caller they have been blacklisted and politely HangUp()
;
; ${ARG1} = CallerID of incoming call
;
exten = s,1,MYSQL(Connect connid ${realdb_host} ${realdb_user}
${realdb_pass} ${realdb_db})
exten = s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ callerid=${ARG1} and blockenabled = 1)
exten = s,3,MYSQL(Fetch fetchid ${resultid} blacklistid)
exten = s,4,MYSQL(Clear ${resultid})
exten = s,5,MYSQL(Disconnect ${connid})
exten = s,6,GoToIf($[”${blacklistid}” = “”]?7:fail,1)
exten = s,7,NoOp(${blacklistid})
; If the callerid is listed in the database, then send to blacklistednumber
; context
;
exten = fail,1,NoOp(${blacklistid})
exten = fail,2,GoTo(blacklistednumber,s,1)
[blacklistednumber]
; This is where a call will land if the macro-checkblacklist decides that
; the number should not be allowed to dial the company.
exten = s,1,Wait(2)
exten = s,2,Playback(privacy-you-are-blacklisted)
exten = s,3,Zapateller()
exten = s,4,HangUp()
On Wed, Feb 25, 2009 at 3:40 PM, Elliot Murdock murdo...@gmail.com wrote:
Hello Everyone!
According to voip-info.org the correcy syntax for the realtime function
is:
REALTIME(family|fieldmatch[|value[|delim1[|delim2]]]) on read
REALTIME(family|fieldmatch|value|field) on write
It seems from the syntax that it is only possible to retrieve a full
row according to the value of only of column. This translates in SQL
language as Select * from family where fieldmath = value.
Is there any way to have more control over the realtime function?
Also, regarding the MYSQL function, I only saw documentation to query
a database. Is there any way to update a database using that
function?
Thanks!
Elliot
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***
Forrest Beck
IAXTEL: 17002871718
jonforrest.b...@gmail.com
http://www.shift8.biz
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Re: [asterisk-users] AGI problem using mono (.Net)
Suggest, Use .net to do an web services and use curl+agi scripts to integrate your solutions. Regards, Luis Morales On Wed, Feb 25, 2009 at 6:37 PM, Douglas Mortensen d...@impalanetworks.com wrote: Hello. I have a software developer creating a .Net / mono program to use as an AGI script. We are having problems getting it to stream files. From what we can tell, it is talking to asterisk correctly when called from the dial plan. Its stderr output goes to the asterisk console. But asterisk doesn't give any indication that it receives the STREAM FILE command. Asterisk simply quickly executes the program and moves to the next step of the dial plan, as though it didn't receive any commands from the program. We know it is running, and outputting its results, because we have called it from within a bash script, and in doing so, I set the script to output stdout to a txt file for testing (like this /var/log/asterisk/querylog). When we do this, the file does end up with the first line showing STREAM FILE filename. We're at a bit of a loss as to what's going on. We have checked filenames and are pretty sure that there are no typos and that the files are there. Further, I have a perl agi script using asterisk::agi that also does a STREAM FILE which runs without any problem. In our dial plan, my perl script runs, gets data from the user via the keypad, puts it in a channel variable, then exits, and his AGI script is immediately called as the next step of the dial plan receiving the channel variable as an argument. It seems that there are not as many out there using mono / .net with AGI. The few examples we've found online are a bit dated. Any help would be greatly appreciated. Thanks much! - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCP, Security+ Linux+, Network+, A+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 ___ -- 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 -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- 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] AGI problem using mono (.Net)
On Wed, 25 Feb 2009, Steve Edwards wrote: The AGI interface (is that redundant?) can be summarized as: 1) Asterisk sends a bunch of cruft (the AGI environment variables) to your program's STDIN. 1a) Your program must read all of the AGI environment variables. 2) Your program sends a request to Asterisk via STDOUT. 3) Asterisk sends a result to your program via STDIN. 4) Your program does something else. 5) go to step 2. It's very simple, but not very forgiving. If you output anything to STDOUT that is not expected, you're hosed. It is possible to write multi-threaded AGIs (eg, play a file while you are waiting for an answer from your credit card processor), but you can only have 1 request active (you've issued the request and you haven't received a result yet) at a time. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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
