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Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

Dear Members, I have been searching for a package in R which can handle multiple seasonality suggested by taylor(2003). It will be great help if anybody has used this on R before (i.e. which package). Thanks in Advance. Best Regards Atul Malik [[alternative HTML version deleted]]

...is NULL [Aug 8 11:51:53] WARNING[19314] chan_sip.c: No SIP tech_pvt! Fixup of SIP/XXXX-00000404 failed. [Aug 8 11:51:53] WARNING[19314] channel.c: Fixup failed on channel SIP/XXXX-00000404<MASQ>, strange things may happen. Does anyone know what this warning means? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

I''ve looked for DIFF tools for OSX but haven''t found anything good yet. Tried guiffy (http://www.guiffy.com) but it''s very slow and the folder comparison doesn''t indicate a change until you drill down to the level with the difference which kinda defeats the purpose. I''m missing TortoiseSVN and the diff tool that comes with it. Any suggestions for

...operand and its position in the instruction (*myitr)->setOperand(k,*vmitr); // Replacing it with the other operand value } } } The problem is that it goes into the end loop. ps. *vsitr and *vmitr are arguments of different call instructions but pointing to the same values. Thanks -- Abid M. Malik ****************************************************** "I have learned silence from the talkative, toleration from the intolerant, and kindness from the unkind"---Gibran "Success is not for the chosen few, but for the few who choose" --- John Maxwell "Being a good person do...

Hi Is there a way I can stop Manager logged on/off messages from going to the console/logs without losing all the other information I need? Regards Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Hi all, When a user make a call to Asterisk, and when user hang up the call at any point of the conversation,? Asterisk will stop Diaplan intermediately. At this situation,? Are there any way to make? Asterisk continue execute the Diaplan ?, so Asterisk can do something like that delete temporary file, .. etc. Thanks in advance, Giang -------------- next part -------------- An HTML

It there a native asterisk dialplan function which will tell me the position of a specific character in a given string? eg if I wanted to find what position the '@' was at in ${SIPURI} Thanks in advance Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M15...

Is it possible to read jpeg files into R? If yes please guide, Thanks.. I tried to search many time but failed to do. Thankis in advance.. with Best Regards, Malik Shahzad Visiting Researcher National Institute of Informatics (NII) Tokyo, Japan Doctoral Student Asian Institute of Technology (AIT) Bangkok, Thailand +66-8-7676-5616 [[alternative HTML version deleted]]

...int(llvm::errs(), false); errs()<<"\n"; } -----> How can I convert the arg for binary comparison(== etc.)? If I am correct, it is not a string. If the argument is "i32 1", Is there a way to access the content value "1" directly? Thanks -- Abid M. Malik ****************************************************** "I have learned silence from the talkative, toleration from the intolerant, and kindness from the unkind"---Gibran "Success is not for the chosen few, but for the few who choose" --- John Maxwell "Being a good person do...

Hi I'm using asterisk 1.8 Does anyone know how to use the queue reload command. The built in help doesn't really help. queue reload {parameters|membe Reload queues, members, queue rules, or parameters Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -------------- next part ----...

Hi Is there any way to set the presence state of a peer to in-use in asterisk 1.8? The idea is to integrate DND buttons on phones to BLF. Regards -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -------------- next part ----...

Hi I have some question about mysql realtime. 1) Anyone know exactly if there is a specific order to declare sip table column for realtime ? In which file can I find that order ? 2) In my extconfig.conf, [settings] are : sipusers => mysql,general,siptable sippeers => mysql,general,siptable so means that I use realtime dynamic exactly ? Is it normal if some parameters from sip.conf still

I have to do user authentication for a project i''m working on. The trick is they want me to use the existing LDAP database for it. Does anyone know how to authenticate using LDAP in RoR? ~Rahul

...does an attended transfer of incoming call to extension B I'm finding that the recording only lasts up to the point that the transfer is made. Is this the correct behaviour? Is there any way I could make this inbound call into a single continuous recording? Thanks in advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

Does it mean we can not dereference the Value variables? Value *val = some operand of an instruction; Value *val2= some operand of another instruction; I am trying to rewire the operand values of an instruction using: *val = *val2; It seems that this is not allowed. Thanks, -- Abid M. Malik ****************************************************** "I have learned silence from the talkative, toleration from the intolerant, and kindness from the unkind"---Gibran "Success is not for the chosen few, but for the few who choose" --- John Maxwell "Being a good person do...

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > Looks like your phones do not support it. And it is a very common feature. I think so... Maybe I can write a little program running on my PC to receive a message from Asterisk if someone calls the other phone... I'll think about that... Or maybe is...

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > Look up fop2 Thank you very much, but I prefer a standalone application, if it's possibile... Any other suggestion? Thanks Luca Bertoncello (lucabert at lucabert.de)

Hi I know this is far from best practice but is it possible to authenticate a sip peer on the IP address it's coming from so that it doesn't need to use a UN and Pass? Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062

...ll the channels which are in use at that time are simply cut off or frozen. The only way for us to get everything back to normal is via a hard restart of asterisk. If you have seen this behaviour or know of a bug which is associated to this can you advise, or point us at a fix. Thanks -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062