search for: g711ulaw

Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I already have t38pt_udptl=yes in sip.conf Thank you.

...e GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've set the all codec option to PCMU only. Due to this I'm facing issue. Any solution for this problem, please let me know. Regards, Keshav Regards, Kesh " Lets change the future...lets change the world." ------------------...

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very bad. If more than one access the conference room it starts to blip real badly. Thots, ideas greatly appreciated. -- respectfully, Joseph ------=============

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be...

...reinvite=no ; [ccme-outbound] type=friend host=10.5.7.130 qualify=yes context=from-ccme trustrpid=yes sendrpid=yes allow=all canreinvite=no dtmfmode=rfc2833 And, in CME: ----------------- dial-peer voice 200 voip session protocol sipv2 incoming called-number 2155551212 dtmf-relay rtp-nte codec g711ulaw no vad ! dial-peer voice 101 voip description softphones 4-N destination-pattern 4[0-9] monitor probe icmp-ping session protocol sipv2 session target dns:sylvester.home.misty.com dtmf-relay rtp-nte codec g711ulaw no vad ! sip-ua no remote-party-id registrar dns:sylvester.home.misty.c...

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

I see a lot of comments but no actual show runs. Can someone post a 7206 config. I am having a dickens of a time getting calls to pass. I currently have the following loaded. Cisco IOS Software, 7200 Software (C7200-IK9O3S-M), Version 12.3(8)T6, RELEASE SOFTWARE (fc2) Thanks !!! Jerry -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 /

...: pbx.c:1318 pbx_extension_helper: Cannot find extension context 'default' Which is correct, meaning the context declaration is not being respected. ------ dial-peer voice 101 voip destination-pattern 10. session protocol sipv2 session target ipv4:10.0.0.133 dtmf-relay rtp-nte codec g711ulaw no vad ------- My bad or something else ?? TIA, Nathan. Here is a sip debug for that peer: Sending to 10.0.9.1 : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.0.9.1:19206 Found description format PCMU Found description format telephone-eve...

...;--------------------> [test server that plays .wav file then DTMF digits] -----PSTN-------[Asterisk]-----SIP----[Cisco 7960] ----<------------------------------------call setup in this direction ---------<---------------< ---------------< The Asterisk is set for DTMF=inband , codec g711ulaw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030812/79698d97/attachment.htm

...his phone to download the SIP firmware? Best, Max -- Max Clark max [at] clarksys.com http://www.clarksys.com -------------- next part -------------- P0S3-06-3-00 -------------- next part -------------- # SIP Configuration Generic File (start) image_version: P0S3-06-3-00 preferred_codec: g711ulaw #preferred_codec: g729a # Proxy Server proxy1_address: "10.6.0.223" proxy2_address: "" proxy3_address: "" proxy4_address: "" proxy5_address: "" proxy6_address: "" # Line 1 Settings line1_name: "Cisco7960"...

...* such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern 9999$ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also tried using rtp payload-type nte to adjust the nte port value to 101 versus 100. SIP Notify doesn't work, cisco-rtp doesn't work. I have tried every possible dtmfmode= option [inband, rfc2833, or info] within this SIP device. Toggling session transport udp, no vad,...

Hi all, A Cisco ATA186 configured with g711ulaw, NAT=yes and canreinvite=yes, made calls through Asterisk to a Cisco 5300 gateway out to a PSTN line with errors as follows: -- Executing Dial("SIP/ata186-c1cf", "SIP/29086988@110.100.231.2:5060|30|r") in new stack -- Called 29086988@110.100.231.2:5060 Apr 15 16:11:22...

...hin the asterisk work perfect, there is not problem. However, the calls that are from Asterisk to the PBX (or viceversa) have a kind of noise every three seconds or so. I have tried different codecs without success and I don't know what else to try. In this moment I only count with the codecs g711ulaw and g711alaw in the router, I don't think bandwidth is a problem in this moment. This is the output while a call is been used: ---- cut here ---- VoIP*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message X.Y.Z.K 2. 83...

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.messenger.yahoo.com

...t the SIP phone to ring - we answer and can hear the originating party - but the SIP softphone is not able to transmit DTMF or audio back to the PSTN... I'm not sure if this is an issue w/ converting the signal in asterisk i.e. SIP to H323 -- or if a problem in the codec or what? The codec is G711uLaw.. Help - thanks Robert A. Huddleston, KF4BYY Cavalier Telephone LLC. 804.422.4401 <mailto:rhuddleston@cavtel.com> rhuddleston@cavtel.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/200409...

...ce modem no cdp enable voice-port 3/0:D bearer-cap Speech dial-peer voice 1 pots incoming called-number 21255512[00-50] direct-inward-dial ! dial-peer voice 100 voip destination-pattern 21255512[00-50] progress_ind setup enable 3 session protocol sipv2 session target ipv4:10.10.10.10 codec g711ulaw no vad ! dial-peer voice 1000 pots destination-pattern .......... port 3/0:D sip-ua retry invite 4 retry response 3 retry bye 2 retry cancel 2 sip-server ipv4:10.10.10.10 line 1/00 1/59 no flush-at-activation no modem InOut transport input all line 2/00 2/59 no flush-at-activation no...

...that it can handle 8 T1s. Half of the calls coming into the system will be routed to SIP extensions (with transcoding). The machine we have in our disposal is a new dual Xeon 3.2gHz server with 2g of ram and an dual 1000mb nic. Voice will be coming in from the PSTN (through 2 quad digium cards) in g711ulaw, and most of the time will be doing IVR with dtmf detection. Half of the calls will eventurally got routed to sip extensions and the requirement calls for keeping the bandwidth to a minimum; so we force all the sip extensions to only use g729. Is this machine capable of meeting the requirement? Thn...

....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... session target ipv4:192.168.1...

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it is doing it very badly. I get stuttered, choppy speech which sounds like it's being played back at half speed. My first question is why is Asterisk attempting to do codec translati...

Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about