similar to: Snome-200 with Asterisk

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

Hi all, Is there a possibility to set the codecs Asterisk will choose in the dialplan ("exten=>" statements or their contexts) instead of sip.conf? My problem is that I connect my SIP phone with several providers (Nikotel, Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers offer the same set of codecs. I'd like Asterisk to use the same codec for the

hi! I've got cisco 7960G working with * box. Calls could be Blind Xfered through the phone but not the supervised transfer( Message on the phone: Transfer failed). Even when I put the caller on hold and resume it later, I can't hear the other side but the otherside can hear me. (It shows as the line is connected though. Yet the respective caller entry blinks.) Any suggestions most

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

hi! I wanna do some arithmatic operations (addition and substraction -integer operation) inside extensions.conf. Is there a simple way to do this. If I do yy = ${xx} + 1 // say "xx" is initialized to '0' the resulting "yy" will show "0 + 1" Obiviously not the result I need. Any help !!!!! denzel.

hi! I'm looking for a robust hardware IP phone which supports SIP protocol inorder to implement a call centre. Have anyone used CISCO SIP phones (eg:- 7960G ) with asterisk. From what I know these CISCO IP phones are very robust and feature rich. Yet I'm nervous whether * don't like CISCO at all. Thoughts are most welcome. denzel. -------------- next part -------------- An HTML

hi! I've got a asterisk system running with around 50 per calls per minute. I've connected * to internal pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have to live with or is it a bug in CVS code ? denzel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32). I use a context [capi-in] for icoming ISDN calls: ====== [capi-in] ; Faxe fuer Ruben exten => 12345,1,Macro(faxin,ruben.roegels at jumping-frog.org,${EXTEN}) ====== My macro for the fax receiving looks like that: ====== [macro-faxin] ; Faxe ; ARG1 =

Hi, I am trying to parse a date (that is read in as a factor) and add it to a dataframe. The length of the parsed date is shorter than the length of unparsed date and I therefore cannot add it to the dataframe: > x [1] 20030807 20030807 20030807 20030808 20030809 20030809 20030809 20030809 [9] 20030808 20030808 20030809 20030808 20030808 20030819 20030819 20030821 . .

Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send

Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. I have the following in sip.conf : dtmfmode = rfc2833 which works well for Grandstream, Yealink and Cisco phones. But not for Snom. Snom support tells me I should use SIP info. Is it possible to have something like this : dtmfmode = rfc2833, info ?? Because

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hi, unfortunately the pbx control sequences which I have to use within asterisk (chan_h323) aren't the same as predefined in our snome phones. Now I found in an old message in the mailing list archive, that the control sequence for direct transfer is just #. Is there somewhere a list with further pbx control sequences (consultant transfer, ...)? Thanks for any hints! Roger.

Hi, I don't know quite how to ask this question, because my knowledge is so limited at this time. I have an h323 phone that I am trying to use to do VOIP to phones on the PSTN. I want to sign up for a service and not have it go out my POTS line. I do have a Quicknet Line jack in my RH 9 box and it is fully confiugred. I have downloaded the latest drive from openh323.org and installed it

Good day all This is what i got off the net about queues and agents "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option above). SIP transfers result in the Agent remaining affiliated with the call until its eventual termination, preventing that agent from being offered another call." We have a snome

hi! I'm trying to get zapras working in GSM csd network. Whenever a dialup call is initiated from the mobile to the * gateway the following appears in the log and zapras terminates. Phone gives the error dialup not answered. ==> /var/log/messages <== pppd[2310]: Plugin zaptel.so loaded. pppd[2310]: Zaptel Plugin Initialized pppd[2310]: Using zaptel device 'stdin' pppd[2310]:

Hello all! I have a few problems with Snom 320 phones: Problem A - Transfer out of Queues: We have a call center with some Snoms. We are using Queue and AgentCallbackLogin. As we run * 1.2.7.1 an agent could transfer a call out of the queue using the hold and transfer buttons on the Snom. This might have been the wrong way to do it all the time I found out later, but it worked. Now we upgraded

Hello! I have finally ordered some Asterisk hardware: the TDM DevKit. However, I want to use VoIP phones (or possibly adapters) for remote users. I would like to get some suggestions on which phones to buy. I'm hoping that some of you with real experience might be able to help me out! Here are the features that are important to me: * While these phones are initially going to be

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Instead of using a PCI card is it possible to use an outside SIP service for "CO" lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/33b131e3/attachment.htm