Hi all, I am an MTech student and currently working on a project on GSM air interface. I am making use of Asterisk soft PBX. I am stuck at a point regarding this. As far as I understood from the available Asterisk documentation that Asterisk can easily plug into it the various programming interfaces and different codecs in it can seemlessly talk to one another. Asterisk has a codec translator API for GSM. Is it possible to make Asterisk directly communicate with a GSM air interface module thourgh the GSM codec API ? That means that i would be making a call from IP phone that wil be routed through the asterisk to the GSM interface. If it is possible, where should i make the necessay changes to enable this interworking. Kindly help. Any kind of suggestions are welcome. Renu Rangnekar
On Wed, 8 Sep 2004 18:58:11 +0530, Renu Rangnekar <rrenu@cedt.iisc.ernet.in> wrote:> As far as I understood from the available Asterisk > documentation that Asterisk can easily plug into it the various programming > interfaces and different codecs in it can seemlessly talk to one another. > Asterisk has a codec translator API for GSM. Is it possible to make Asterisk > directly communicate with a GSM air interface module thourgh the GSM codec > API ? That means that i would be making a call from IP phone that wil be > routed through the asterisk to the GSM interface. If it is possible, where > should i make the necessay changes to enable this interworking.It will take a lot more than the codec translator and "changes" to make Asterisk talk to a GSM BTS. You would have to implement a significant part of the GSM MAP protocol and an SS7 stack. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.
hi all i have asterisk on a TE410P card running on 2 E1s. E1(a) and E1(b) Following is the call flows 1.user call from POTS to E1(a) 2. asterisk will authentic user ANI and once authentic is clear it will play DTMF A tone 3. user key in destination number 4. asterisk uses LCR to find which Telco is the cheapest 4. asterisk pick up E1(b) and dial a local Telco gateway number 5. waits for DTMF A from Telco and send the destination number to Telco 6. asterisk connects user and destination user user-->POTS-->E1(a)-->asterisk-->E1(b)-->E1-->Telco-->destination the only problem with the above flow is, when E1(b) calling Telco E1. two things happen frequently 1. connection takes more than 10 secs 2. not all calls are successful appreciate if any one has done this before and can assist. willing to outsource if anyone has the solutions rdgs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041101/0f2e24a0/attachment.htm