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...ssages when checked? It also keeps insisting that the user's voice mailbox is full and can't store more messages even if I clear/rebuild the /var/spool/asterisk/voicemail stuff. I've tried falling back to voicemail.conf entries from realtime voicemail with the same result. Thanks, Vahan -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050714/7500dddb/vahan.vcf

Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten => _2XXXXXXXX,1,Dial(SIP/${EXTEN}@voiprovider) Now, with Asterisk2Billing would be something like this? exten => _2XXXXXXXX,1,Answer exten => _2XXXXXXXX,2,Wait,2 exten => _2XXXXXXXX,3,DeadAGI,a2billing.php exten => _2XXXXXXXX,4,Wait,2 exten =>

http://software.newsforge.com/software/05/07/06/1235211.shtml?tid=150&tid=147&tid=136 http://www.theregister.co.uk/2005/07/06/eu_bins_swpat/ -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/357c6cce/vahan.vcf

...is no password. Linking the username to a specific ip could be some kind of security measure so noone could masquarade as that username. what i want is something like what we have in sip.conf: [username] host=dynamic ; <--- this one, we can put IP here instead of 'dynamic' regards, Vahan

...le extensions? In my case extensions are the same as the sip usernames, while as per http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence exten => 1234,hint,SIP/1234 works, exten => _1XXXX,hint,SIP/${EXTEN} doesn't. Not sure if I can even use ${EXTEN} here... Any hints? Vahan -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050907/d97d24b6/vahan.vcf

...sales/support emails? here is the snip from my extensions.conf, anything wrong? exten => _3741XXXXXX,1,Dial(SIP/1111${EXTEN:4}@x.y.z.x,60) exten => _3741XXXXXX,2,Congestion exten => _3741XXXXXX,3,Hangup exten => _3741XXXXXX,102,Congestion exten => _3741XXXXXX,103,Hangup regards, Vahan

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

...o this seems to be a problem with chan_sip. I'm hesitant to post the long sip debug outputs to the mailing list to conserve the bandwidth. More info and sip debugs are available at http://bugs.digium.com/bug_view_page.php?bug_id=0003726 Is there anyone else with the same problem? regards, Vahan -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050307/8165143a/vahan.vcf

...wondering if anyone has successfully configured two lsi/3ware 9750-4i series controllers for multipathing under CentOS 5.7 x86_64? I've tried some basic setups with both multibus and failover settings, and had repeatable filesystem corruption over a iscsi(tgtd) or nfs3 connection. Any ideas? Vahan

...5e9ad727e87a6f", algorithm=MD5 ..asterisk rejects the call with Failed auth on 010527911@195.250.77.70 :( Is there a limitation in Asterisk and it uses the "From" address as the auth user? This seems buggy.. I'll send the full debugs off-list if someone is interested. regards, Vahan -------------- next part -------------- A non-text attachment was scrubbed... Name: vahan.vcf Type: text/x-vcard Size: 287 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050726/39e1d848/vahan.vcf

Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- Tomislav Parcina tparcina#lama.hr

We are using the following to record conversations. exten => _1XXX.,1,SetVar(CALLFILENAME=call_to_${EXTEN:1}_dated_${TIMESTAMP}) exten => _1XXX.,2,Monitor(wav,${CALLFILENAME},m) exten => _1XXX.,3,Dial(IAX2/4506:zj5S3A5a@nl.voipgate.nl/${EXTEN:1}) exten => _1XXX.,4,Congestion exten => _1XXX.,104,Congestion This was working previously to record both sides of the conversation but now

...see those in the embedded website's codec priority page. Is there something more to be done to enable g723.1/gsm codecs? P.S. 3502A is also affected by the registration bug, if you connect a phone to TEL2 jack and call someone, everything goes by the TEL1's account/password... regards, Vahan Yerkanian

...; 451,3,VoiceMailMain(${CALLERIDNUM}@sip) exten => 451,4,Wait(1) exten => 451,5,Hangup exten => 452,1,Answer exten => 452,2,Zapateller exten => 452,3,Voicemail() exten => 452,4,Wait(1) exten => 452,5,Hangup The last extension, 452, is the one that doesn't work. regards, Vahan

I've searched for a few days now without finding an answer. The release notes for version 1.0.5.16 of the Grandstream firmware says it supports attended transfer using replace but the docs haven't been updated so I can't work out how to enable it, or whether it should Just Work. I'm currently using the # attended transfer patch for * but would like to get back to using the

Where I can find the firmware for the Wellgate 3804 ? The files are: - 2m4sipfxo.103 - 4fxosip.103 I don't have a password to pick up it at the welltech site. Kind regards, Miguel

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone used this board as an Asterisk server? http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y I'm mostly interested about the possible compatibility issues this board may have with the AEX800 card.

asterisk-users-bounces@lists.digium.com wrote: > Is there anyone else with the same problem? Yes, we've seen the same problem. We have found a work around, but I'm unable to to look into it today. -- Andreas Sikkema Rits tele.com Van Vollenhovenstraat 3 3016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different