search for: g711alaw

...STN BRI1 incoming called-number 2012345.. no digit-strip direct-inward-dial port 1/0/1 ! dial-peer voice 30 voip description INBOUND CALLS VOIP ASTERISK destination-pattern 2051860.. session protocol sipv2 session target ipv4:y.y.y.y:5060 session transport udp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 40 voip description OUTBOUND CALLS VOIP CARRIER destination-pattern .+ session protocol sipv2 session target ipv4:x.x.x.x:5060 session transport tcp dtmf-relay sip-notify codec g711alaw no vad ! dial-peer voice 50 pots tone ringback alert-no-PI description OUTBOU...

...ource connected at a E1 Voice Link. I want that all call incoming on the cisco 5300 are sent to Asterisk and all Asterisk outgoing call are sent to Cisco AS5300. Actually, i configure the AS5300: isdn switch-type primary-net5 ! voice service voip sip ! voice class codec 400 codec preference 1 g711alaw codec preference 2 g729r8 codec preference 3 g723r63 codec preference 4 g711ulaw ! voice class codec 500 codec preference 1 g729r8 codec preference 2 g723r63 ! controller E1 0 framing NO-CRC4 pri-group timeslots 1-31 description E1 Beta-Test interface Serial0 no ip address shutdown clo...

I wouldn't expect you to be using RFC3389 if your using A-law, can you include your IOS version and IOS config file ... I have not specified any allow's or disallow's in my * config for the codecs with my 5300, I also use Cisco 79xx phones and I use the option within the phones config file to select the preffered codec and when I change this to G.729/A-law/U-law all works perfectly

...nopassword username ecn nopassword ! ! resource-pool disable ! ip subnet-zero no ip source-route no ip routing ! isdn switch-type primary-net5 voice rtp send-recv ! voice service pots ! voice service voip sip bind all source-interface FastEthernet0 ! voice class codec 723 codec preference 1 g711alaw ! voice class codec 1 codec preference 1 g711alaw ! voice class codec 7 codec preference 1 g711alaw ! ! ! ! ! ! ivr prompt memory 16384 files 1000 fax interface-type fax-mail mta receive maximum-recipients 0 call-history-mib max-size 500 dial-control-mib max-size 1200 ! controller E1 0 ds0-gr...

hello thanks Dmitry for your useful hints. i enable debug and solve my problem:). it was codec compatibility problem. but it is so strange; if i set codec g711alaw in cisco router and asterisk, i have the mentioned problem but if i set codec to transparent in cisco router, every thing will be ok. is there any difference between g711 codecs which cisco and asterisk utilize? On Wed, May 6, 2015 at 11:56 AM, Dmitry Melekhov <dm at belkam.com> wrote: &gt...

I've previously been using G711alaw on both the AS5300 and the phones but feel the need for a less bandwidth hungry codec for those users that are connected behind ADSL and so was investigating G.729 but .. Firstly I found that on my AS5300 I have either G.729r8 or G.729br8 and on the 7940 phones I have G.729a, I'm not sure whic...

Setup: Asterisk server in NY. Cisco 7960 IP Phones in NY and London. Dedicated T1 from NY to Ldn. T1: Latency - 100ms Qos applied No errors Default codec on Ldn IP Phones = g711alaw Default codec on NY IP Phones = g711ulaw Both codecs allowed on each phone. Issue: Calls on IP Phones from NY to London hear clicking noise on NY end. Anyone experienced something similar or can offer some assistance? Thanks, Taf.. Send instant messages to your online friends http://uk.messenge...

...1 type=friend host=192.168.13.1 nat=yes canreinvite=no dtmfmode=rfc2833 qualify=no secret=cisco cisco 2811 config: ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 cause-code legacy sip ! voice class codec 99 codec preference 1 g711alaw codec preference 2 g711ulaw (SNIP) ! dial-peer voice 1 voip answer-address 1... destination-pattern 1... voice-class codec 99 voice-class h323 1 session target ipv4:192.168.100.44 dtmf-relay h245-alphanumeric no vad ! dial-peer voice 2 voip answer-address 2... destination-pattern 2......

...ecode/encode the Q.sig. The Nortel should be defined to send and receive names via Q.sig. The definition fragments on Cisco are: isdn switch-type primary-qsig (so it will use Q.sig signalling). ... voice service voip qsig decode (This sends names out via Q.sig) fax protocol pass-through g711alaw sip .... controller E1 0/0/0 pri-group timeslots 1-31 ... interface Serial0/0/0:15 (This is for E1 PRI). no ip address encapsulation hdlc isdn switch-type primary-qsig isdn overlap-receiving isdn not-end-to-end 64 isdn incoming-voice voice isdn supp-service name calling (This receive...

...1/00 CSMV6 0000-0005 0.10.2.2 N/A ios-bundled default [...] Router#show conf [...] spe country t1-default isdn switch-type primary-ni ! voice hunt user-busy voice call send-alert voice call convert-discpi-to-prog voice rtp send-recv voice service voip fax protocol pass-through g711alaw h323 sip bind all source-interface FastEthernet0/0 controller T1 3/0 framing esf linecode b8zs pri-group timeslots 1-24 interface Serial3/0:23 description T1 to CLEC no ip address load-interval 30 isdn switch-type primary-ni isdn incoming-voice modem no cdp enable voice-port 3/0:D bea...

I have several 7960 phones with SIP image (7.3) and Asterisk 1.0.1 on FreeBSD. When I have 2 active SIP calls on the 7960 phone there are no available lines for additional calls. I tried to configure 2 lines to the same SIP server but it's still limited to 2 calls. How to utilize all lines? -- Called user -- SIP/user-acc6 is ringing -- SIP/user-acc6 answered SIP/x.x.x.x-09a9a000 --

...rvice timestamps log uptime service password-encryption ! hostname gw1 ! boot-start-marker boot system tftp mc3810-a2isv5-mz.123-10a.bin 192.168.5.104 boot-end-marker network-clock base-rate 56k no aaa new-model ip subnet-zero ! voice class codec 10 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 4 g729r8 codec preference 6 g729ar8 ! ! no voice confirmation-tone ! controller T1 0 mode cas framing esf linecode b8zs ds0-group 1 timeslots 1-4 type e&m-wink-start fdl both ! controller T1 1 mode cas framing esf linecode b8zs ds0-group 1 timeslots 1-4 type e&m-...

...eFax(${INCOMING_FAXFILE}) sip.conf [general] canreinvite=no t38pt_udptl=yes disallow=all allow=alaw context=fax-in The CISCO peer configuration: dial-peer voice 6 voip destination-pattern 88T session protocol sipv2 session target ipv4:10.100.0.51 session transport udp dtmf-relay rtp-nte codec g711alaw fax-relay ecm disable fax nsf 000000 fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback none no vad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/af07d09e/attachment-0001.htm

Hello, I'm trying to receive faxes with asterisk. My configuration is like this: PSTN fax -> ISDN -> Cisco router with VoIP module -> Asterisk When I try to send a fax from PSTN fax I got the standard fax signal, Asterisk starts rxfax application and then call ends and there is no tif anywhere. On the fax display there is still one message: Calling... Part of my extensions.conf:

...[general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip> dtmf-relay rtp-nte codec g711alaw no vad ! When I try to make a call, cisco shows codec g711alaw, but asterisk shows codec g729A (i have the licenses) and there is no audio. When I try disallow=g729, the same occurs, but this time asterisk shows codec gsm. The only way to make a call is allowing only alaw. But this...

...fxroute ! logging queue-limit 100 enable secret enable password ! clock timezone GMT 0 ip subnet-zero no ip routing ! ! ! ip audit notify log ip audit po max-events 100 ! ! ! ! ! voice rtp send-recv ! voice service voip sip ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g711alaw codec preference 3 gsmefr codec preference 4 gsmfr ! ! ! ! ! ! ! no voice hpi capture buffer no voice hpi capture destination ! ! mta receive maximum-recipients 0 ! ! ! ! interface Ethernet0/0 ip address xxx.xxx.xxx.yyy 255.255.255.0 no ip route-cache no ip mroute-cache full-duplex no cdp en...

..."32766" nat_received_processing: "0" # Proxy Registration (0-disable (default), 1-enable) proxy_register: "1" # Phone Registration Expiration [1-3932100 sec] (Default - 3600) timer_register_expires: "3600" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # Enable VAD (0-disable (default), 1-enable) enable_vad: "0" # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: "1"...

...i believe this problem is a codec problem as far as i can see we use ulaw across the board, the 5300 currently supports 12 different codecs however asterisk only like too work with ulaw or alaw it tends to not except the call if the other codecs are used. clear-channel Clear Channel 64000 bps g711alaw G.711 A Law 64000 bps g711ulaw G.711 u Law 64000 bps g723ar53 G.723.1 ANNEX-A 5300 bps g723ar63 G.723.1 ANNEX-A 6300 bps g723r53 G.723.1 5300 bps g723r63 G.723.1 6300 bps g726r16 G.726 16000 bps g726r24 G.726 24000 bps g726r32...

...with the cisco I hear nothing, I'v tested codecs ulaw and alaw but the both do the same. This is my cisco's config dial-peer voice 20 voip destination-pattern 02322663910 translate-outgoing called 20 session protocol sipv2 session target ipv4:200.85.96.230 dtmf-relay cisco-rtp codec g711alaw ! translation-rule 20 Rule 0 ^02322663910 1234 ! Any ideas?? Luciano Ramos CCNA - MCP Jefe Depto. Internet TelViso 02320-470300

Hello. I have been searching the archives for a simple, clear listing of the available codecs with total size, plus the data and overhead sizes. Does anyone have this handy, and can it be added somewhere, even the wiki. Regards...Martin -- The system will be down for 10 days for preventive maintenance.