[asterisk-users] How to put your call on hold with asterisk Dialplan

Hi List, Is it possible to put your call on hold after bridge with another call ? I want to use asterisk dialplan not softphone button or IP-phone device. If yes then how to get back on bridge mode of on-hold call again ? - Thanks and regards Virendra Bhati +91-9172341457 Asterisk

Re: [asterisk-users] Skype for Asterisk - RIP

On Tue, May 24, 2011 at 10:50 PM, Matt Darnell wrote: > "We expect that users of Skype for Asterisk will be able to continue > using their Asterisk systems on the Skype network until at least July > 26, 2013. Skype may extend this at their discretion." It's widely believed. However, it's very pos

[asterisk-users] synway

Dear, do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ? is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -- _ -- Bandwidth and Colocation Provided by http://www.api-di

Re: [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation

> -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of > Jose P. Espinal > Sent: Tuesday, May 24, 2011 10:10 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] How t

Re: [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation

On 05/24/2011 07:20 PM, bilal ghayyad wrote: To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? https://wiki.asterisk.org/wiki/display/AST/Using+Menuselect+to+Select+Asterisk+Options -- Jose P. Espinal http://w

Re: [asterisk-users] How to enable the addon in the Asterisk 1.8 compilation

On 05/25/2011 01:20 AM, bilal ghayyad wrote: Hi All; To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? make menuselect and enable what you would like to be build. Regards, Patrick -- __

[asterisk-users] How to enable the addon in the Asterisk 1.8 compilation

Hi All; To enable the compilation for the addon that is coming with Asterisk 1.8 when doing compilation for the Asterisk, what should I do? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

[asterisk-users] Asterisk SIP Trunk with CUCM Express, Disable Comfort Noise?

Hi All, I have a sip trunk up and running with a CUCM Express, passing calls fine except for a comfort noise error I'm getting on Asterisk: NOTICE[7520]: rtp.c:788 in process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: x.x.x.

Re: [asterisk-users] Faxing with Asterisk 1.8.4 & T.38

On 05/24/2011 01:07 PM, e...@erols.com wrote: > I have tried faxing to the DID from 2 different fax machines connected to > different POTS lines. One fax machine is a Xerox Workcentre, and the other > is a Brother Intellifax. Can you provide some more information about your > setup? If you wo

Re: [asterisk-users] AstManProxy

On 05/24/2011 03:43 PM, Steve Davies wrote: [snip] Repo updated. I have tried to merge all of the other changes that are "out there" also. Thanks Steve. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www

[asterisk-users] Asterisk + Patton ISDN2e gateway (UK)

I know I asked this some time back, and I got no response then, and neither did someone who asked at the start of 2009 either by the looks of it (other than a reply from me to use a PCI card!!!) However I now have a client who'd bought one of these boxes and installed ISDN2e - against my adv

Re: [asterisk-users] MagicJack quality

On 05/24/2011 02:45 PM, Warren Selby wrote: On Tue, May 24, 2011 at 12:33 PM, Doug Lytle > wrote: Steve Edwards wrote: My archives don't go back that far Mine do. No match on Jack, magic or itntelecom.com

Re: [asterisk-users] MagicJack quality

Warren Selby wrote: A quick google for "magicjack quality C. Savinovich" Interesting, it would appear that my archives are incomplete. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _

Re: [asterisk-users] MagicJack quality

On Tue, May 24, 2011 at 12:33 PM, Doug Lytle wrote: > Steve Edwards wrote: > >> My archives don't go back that far >> > > Mine do. > > No match on Jack, magic or itntelecom.com c.savinov...@itntelecom.com> > > Doug > > http://www.mail-archive.com/asterisk-users@lists.digium.com/msg207540.html A

Re: [asterisk-users] Faxing with Asterisk 1.8.4 & T.38

Eric Contact me off list if you like and we can talk about your t38 fax needs. We may be able to help depending on your location. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "e...@erols.com" Sent: Tuesday, May 24, 2011 2:08 P

Re: [asterisk-users] Faxing with Asterisk 1.8.4 & T.38

On May 20, 2011, at 3:30 PM, Anthony Messina wrote: > On 05/20/2011 01:20 PM, e...@erols.com wrote: >> #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to >> receive faxes via T.38. Sending faxes is not a requirement. Does anyone >> have a working asterisk 1.8.4 configur

Re: [asterisk-users] chan_zap

Dear; I did restart for asterisk and actually i restarted the machine using init 6 but the problem still the same. I run zttool and it gives OK and not REC, also when I pressed F1 for more details I saw that Current Allarms: No Allarms Sync source: Internally clocked IRQ Misses:

Re: [asterisk-users] MagicJack quality

Steve Edwards wrote: My archives don't go back that far Mine do. No match on Jack, magic or itntelecom.com Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -

Re: [asterisk-users] MagicJack quality

On 11-05-24 01:21 PM, Steve Edwards wrote: > If it take the OP (of this thread) 3 years to reply, what does that say about > their product support? Par for the course? :) Leif. -- _ -- Bandwidth and Colocation Provided by http:/

Re: [asterisk-users] MagicJack quality

On 05/24/2011 11:35 AM, A J Stiles wrote: No they didn't. The "original message" to which the spammer was pretending to reply (and in the wrong place, even) never existed in the first place. > On Fri, Mar 21, 2008 at 4:29 PM, Matthew Rubenstein... My archives do

[asterisk-users] Asterisk 1.8.4.1 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.4.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.4.1 resolves several issues reported by the community. Without your help this release w

Re: [asterisk-users] MagicJack quality

On 05/24/2011 11:57 AM, A J Stiles wrote: On Tuesday 24 May 2011, jon pounder wrote: On 05/24/2011 11:35 AM, A J Stiles wrote: Someone asked about the quality of it, he was quoting the hardware specs of a similar device. No they didn't. The "original message" to which the spammer was pretend

Re: [asterisk-users] MagicJack quality

On Tuesday 24 May 2011, jon pounder wrote: > On 05/24/2011 11:35 AM, A J Stiles wrote: > > > Someone asked about the quality of it, he was quoting the hardware specs > of a similar device. No they didn't. The "original message" to which the spammer was pretending to reply (and in the wrong plac

Re: [asterisk-users] MagicJack quality

On 05/24/2011 11:35 AM, A J Stiles wrote: Someone asked about the quality of it, he was quoting the hardware specs of a similar device. I doubt magicjack publishes that kind of detail about theirs. so where is the problem ? Its irrelevant he represents that device commercially. On Tuesda

Re: [asterisk-users] MagicJack quality

On Tuesday 24 May 2011, BroadTel wrote: > Hi all, > Just in case if anyone will be interested in *REDACTED*, a USB to FXS > adapter embedded with SIP softphone. Product specification is as follows: Please refer to the fifth and sixth words of the title of this mailing list. To everyone else, I re

[asterisk-users] Realtime dbase table mods

Hi all, Anyone know if it's possible to force asterisk to use 'my_table' for passwords instead of the 'secret' table? S. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] AstManProxy

On Tue, 2011-05-24 at 10:02 +0100, Steve Davies wrote: > On 20 May 2011 16:16, Ishfaq Malik wrote: > > On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote: > >> On 11-05-20 09:37 AM, Ishfaq Malik wrote: > >> > Do many people use this? > >> > Is it reliable and safe? > >> > >> It may still work, b

[asterisk-users] SMS callfile

Hi, I am looking for tutorial to generate a callfile so that after my program executes, a callfile is generated and pass to asterisk to send to the recipient. Any suggestion? Besides, do you know if there is a web-based GUI to send sms via asterisk? Thanks. CK -- __

Re: [asterisk-users] AstManProxy

On 24 May 2011 10:43, Patrick Lists wrote: > On 05/24/2011 11:02 AM, Steve Davies wrote: > [snip] >> >> I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most >> recent version is on Github, and is not that old. In fact that reminds >> me that I really must upload my latest changes to

Re: [asterisk-users] MagicJack quality

Hi all, Just in case if anyone will be interested in BroadTel UPA-1, a USB to FXS adapter embedded with SIP softphone. Product specification is as follows: Hardware: USB to RJ11 FXS adapter 1 USB port, for computer connection 1 RJ-11 FXS, for phone connection Dimension (L x W x H): 53 x 15 x 28 mm

Re: [asterisk-users] SIP per-call heartbeat?

On 24/05/11 12:08, Tony Mountifield wrote: OK, thanks. Sounds like there was some kind of issue at the ITSP then. I have seen this happen with broken SIP-ALGs in routers too. The ITSP sends the BYE but for some reason a broken SIP ALG will not deliver the packet to the right place. The IT

Re: [asterisk-users] SIP per-call heartbeat?

In article <098fafc6-cc79-49d4-bdf9-95beeb4de...@edvina.net>, Olle E. Johansson wrote: > > 24 maj 2011 kl. 12.19 skrev Tony Mountifield: > > > One of our customers has an Asterisk conference bridge connected to a > > SIP trunk from an ITSP. Yesterday, they had two inbound calls that > > didn't g

Re: [asterisk-users] Sending call to specific IP address

On 23/05/11 22:30, Elliot Murdock wrote: Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is "j...@phone.com " needs to be sent to the IP address 123.456.789.255, not to the IP

Re: [asterisk-users] chan_zap

try to verifiy the zapata telephony interfaces with this command zttool and verify the status of alarms is OK or REC 2011/5/24 James zhu > hi: > yes, it is true, the channel is locked by some tasks. you can please > restart asterisk and zatpel. > > Best regards, > James.zhu > Doing asterisk/P

Re: [asterisk-users] SIP per-call heartbeat?

24 maj 2011 kl. 12.19 skrev Tony Mountifield: > One of our customers has an Asterisk conference bridge connected to a > SIP trunk from an ITSP. Yesterday, they had two inbound calls that > didn't get hung up properly. From the tcpdump SIP trace that we have > running continuously, I can see that

[asterisk-users] SIP per-call heartbeat?

One of our customers has an Asterisk conference bridge connected to a SIP trunk from an ITSP. Yesterday, they had two inbound calls that didn't get hung up properly. From the tcpdump SIP trace that we have running continuously, I can see that no BYE was received by the bridge, and when some hours l

[asterisk-users] issue with asterisk 1.4

Hi List With asterisk 1.4 intermittently I have an issue when i call manually using a hard phone Snom (sip extension) a number X I found another client with another number Y I don’t know if I have any things wrong in my configuration .i have this issue 1/100 Any help will be appreciated.

Re: [asterisk-users] AstManProxy

On 05/24/2011 11:02 AM, Steve Davies wrote: [snip] I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most recent version is on Github, and is not that old. In fact that reminds me that I really must upload my latest changes to my github fork of the project! The last update to Dave T

Re: [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?

On 05/24/2011 10:50 AM, Olivier wrote: 2011/5/24 Kristijan Vrban mailto:vrban.l...@googlemail.com>> http://code.google.com/p/zaphfc/ Hi, Which hardware would be recommended to try this code ? As far as I know you can use any ISDN card with a Cologne HFC-S chip and a _single_ BRI ISDN

Re: [asterisk-users] Sending call to specific IP address

Hello, It seems that the outboundproxy parameter in sip.conf is what is needed. --Elliot On Tue, May 24, 2011 at 12:15 PM, Elliot Murdock wrote: > Hello, > > Thank you for the reply. > > How would one go about adding a SIP proxy IP address in Asterisk for peers? > > The host parameter for a pe

Re: [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?

On 05/24/2011 09:56 AM, Kristijan Vrban wrote: http://code.google.com/p/zaphfc/ Thanks! For the archives I found another site where the zaphfc code from that google site is integrated into the DAHDI source: http://sourceforge.net/projects/dahdi-zaphfc/ Regards, Patrick -- _

Re: [asterisk-users] Sending call to specific IP address

Hello, Thank you for the reply. How would one go about adding a SIP proxy IP address in Asterisk for peers? The host parameter for a peer (in sip.conf) seems to define both the domain URI and the IP address for the next SIP proxy hop (there is an ipaddr parameter, but it seems that only works in

Re: [asterisk-users] AstManProxy

On 20 May 2011 16:16, Ishfaq Malik wrote: > On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote: >> On 11-05-20 09:37 AM, Ishfaq Malik wrote: >> > Do many people use this? >> > Is it reliable and safe? >> >> It may still work, but that code is quite old, and I'm not even sure it's >> necessary an

Re: [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?

2011/5/24 Kristijan Vrban > http://code.google.com/p/zaphfc/ > > Hi, Which hardware would be recommended to try this code ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Joi

[asterisk-users] Grandstream and setvar

Hi all. I'm trying to use the setvar field in my RT sip configuration to store some information about a phone. This works perfectly for my Polycom phones. For the most part, it works for the Grandstream as well. However, when the GS tries to transfer a call, the variable isn't available. For

Re: [asterisk-users] Sending call to specific IP address

23 maj 2011 kl. 23.36 skrev Paul Belanger: > On 11-05-23 05:30 PM, Elliot Murdock wrote: >> Hello, >> >> I am wondering how to send a call to a specific IP address that is different >> than the host of the URI. For example, an invite to the URI is " >> j...@phone.com" needs to be sent to the IP

Re: [asterisk-users] Latest DAHDI/libpri/Asterisk 1.8 & 1x BRI port HFC based ISDN card?

http://code.google.com/p/zaphfc/ 2011/5/23 Patrick Lists : > Hi, > > I would appreciate some advice on the following: how does one use a single > BRI port HFC chipset based ISDN cards with the latest DAHDI, libpri and > Asterisk 1.8? > > For Asterisk 1.4 I would first install mISDN & mISDNuser and